Logic Studio Instruments and Effects
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1 Contents Preface 11 11 14 15 17 Introduction to the Logic Studio Plug-ins Logic Pro Effects and Instruments Soundtrack Pro Effects WaveBurner Effects MainStage Instruments and Effects Chapter 1 19 19 21 Amp Modeling Bass Amp Guitar Amp Pro Chapter 2 27 28 46 46 47 48 Delay Delay Designer Echo Sample Delay Stereo Delay Tape Delay Chapter 3 51 52 53 54 55 56 57 Distortion Bitcrusher Clip Distortion Distortion Distortion II Overdrive Phase Distortion Chapter 4 59 61 62 65 67 68 70 71 72 Dynam
76 78 79 79 Noise Gate Silver Compressor Silver Gate Surround Compressor Chapter 5 83 84 88 88 90 91 96 97 98 EQ Channel EQ DJ EQ Fat EQ Linear Phase EQ Match EQ Single Band EQs Silver EQ Frequency Ranges Used With EQ Chapter 6 99 100 103 107 116 118 120 Filter AutoFilter EVOC 20 Filterbank EVOC 20 TrackOscillator Fuzz-Wah Spectral Gate Soundtrack Pro Autofilter Chapter 7 121 121 123 125 Imaging Binaural Post-Processing Plug-in Direction Mixer Stereo Spread Chapter 8 127 128 128 129 129 133 1
147 148 149 150 Rotor Cabinet Scanner Vibrato Spreader Tremolo Chapter 10 151 151 155 156 Pitch Pitch Correction Pitch Shifter II Vocal Transformer Chapter 11 159 160 161 162 164 167 168 Reverb AVerb EnVerb GoldVerb PlatinumVerb SilverVerb Soundtrack Pro Reverb Chapter 12 169 171 174 178 180 183 184 186 188 189 Convolution Reverb: Space Designer Impulse Response Parameters Global Parameters Output Parameters Envelope and EQ Display Volume Envelope Parameters Filter Parameters Synthesizer Impulse R
Chapter 15 205 206 206 206 207 208 209 210 215 217 219 221 223 224 224 225 225 226 227 EVOC 20 PolySynth Vocoder Basics What Is a Vocoder? How Does a Vocoder Work? How Does a Filter Bank Work? Using the EVOC 20 PolySynth EVOC 20 PolySynth Parameters Synthesis Parameters Sidechain Analysis Parameters Formant Filter Parameters Modulation Parameters Unvoiced/Voiced (U/V) Detection Output Parameters Block Diagram Tips for Better Speech Intelligibility Editing the Analysis and Synthesis Signals Avoiding Son
Chapter 22 285 287 291 293 300 301 302 305 305 315 The LFOs The Envelopes (ENV 1 to ENV 3) The Square The Vector Envelope Effect Processor Using Controls and Assigning Controllers Random Sound Variations Tutorials Sound Workshop Templates for the ES2 323 324 324 326 326 327 328 329 329 330 331 332 333 335 336 337 338 340 340 347 348 349 351 352 353 355 356 357 358 358 EVB3 MIDI Setup Playing Both Manuals and the Pedals Live Keyboard Split Transposition (Octave Range) MIDI Mode The EVB3 Parameters Drawba
Chapter 23 359 359 360 376 377 EVD6 About the EVD6 The EVD6 Parameters Controlling the EVD6 via MIDI A Brief History of the Clavinet Chapter 24 379 379 380 387 390 EVP88 About the EVP88 The EVP88 Parameters Emulated Electric Piano Models EVP88 and MIDI Chapter 25 391 392 394 396 397 398 399 408 425 446 449 450 EXS24 mkII Learning About Sampler Instruments Loading Sampler Instruments Working With Sampler Instrument Settings Managing Sampler Instruments Searching for Sampler Instruments Importing S
511 511 511 512 516 526 527 527 545 Programming: Quick Start Guide Approaches to Programming Basics The Core Engine Creating Basic Sounds Modulations Programming: In Depth Programming Electric Basses With Sculpture Synthesized Sounds Chapter 29 553 554 555 556 557 563 581 590 603 Ultrabeat The Structure of Ultrabeat Overview of Ultrabeat Loading and Saving Sounds The Assignment Section The Synthesizer Section Modulation The Step Sequencer Creating Drum Sounds in Ultrabeat Chapter 30 615 616 GarageBan
Preface Introduction to the Logic Studio Plug-ins The Logic Studio music and audio production suite features a comprehensive collection of powerful plug-ins. These include innovative synthesizers, high quality effect plug-ins, a powerful sampler, and authentic recreations of vintage instruments. This manual will introduce you to the individual effects and instruments—and their parameters. All plug-in parameters are discussed in detail.
Effect category Included effects Dynamic             Adaptive Limiter (p. 61) Compressor (p. 62) DeEsser (p. 65) Ducker (p. 67) Enveloper (p. 68) Expander (p. 70) Limiter (p. 71) Multipressor (p. 72) Noise Gate (p. 76) Silver Compressor (p. 78) Silver Gate (p. 79) Surround Compressor (p. 79) EQ        Channel EQ (p. 84) DJ EQ (p. 88) Fat EQ (p. 88) Linear Phase EQ (p. 90) Match EQ (p. 91) Single Band EQs (p. 96) Silver EQ (p. 97) Filter      AutoFilter (p.
Effect category Included effects Reverb       AVerb (p. 160) EnVerb (p. 161) GoldVerb (p. 162) PlatinumVerb (p. 164) SilverVerb (p. 167) Convolution Reverb: Space Designer (p. 169) Specialized       Denoiser (p. 192) Enhance Timing (p. 193) Exciter (p. 194) Grooveshifter (p. 195) Speech Enhancer (p. 196) SubBass (p. 197) Utility     Down Mixer (p. 199) Gain (p. 200) I/O (p. 201) Test Oscillator (p. 202) The following table outlines the instruments included with Logic Pro.
Soundtrack Pro Effects The following table outlines the effects included with Soundtrack Pro. Note: Effects included in Soundtrack Pro do not feature the extended parameters that are covered in this document. 14 Effect category Included effects Delay  Stereo Delay (p. 47)  Tape Delay (p. 48) Distortion       Bitcrusher (p. 52) Clip Distortion (p. 53) Distortion (p. 54) Distortion II (p. 55) Overdrive (p. 56) Phase Distortion (p. 57) Dynamic          Adaptive Limiter (p.
Effect category Included effects Reverb  PlatinumVerb (p. 164)  Soundtrack Pro Reverb (p. 168)  Convolution Reverb: Space Designer (p. 169) Specialized  Denoiser (p. 192)  Exciter (p. 194)  SubBass (p. 197) Utility  Gain (p. 200)  Multichannel Gain (p. 202)  Test Oscillator (p. 202) WaveBurner Effects The following table outlines the effects included with WaveBurner.
Effect category Included effects Filter     Imaging  Direction Mixer (p. 123)  Stereo Spread (p. 125) Metering      BPM Counter (p. 128) Correlation Meter (p. 128) Level Meter (p. 129) MultiMeter (p. 129) Tuner (p. 134) Modulation            Chorus (p. 136) Ensemble (p. 136) Flanger (p. 137) Microphaser (p. 138) Modulation Delay (p. 138) Phaser (p. 140) Ringshifter (p. 142) Rotor Cabinet (p. 147) Scanner Vibrato (p. 148) Spreader (p. 149) Tremolo (p.
MainStage Instruments and Effects The following tables outline the effects and instruments included with MainStage. Note: MainStage is a live application, and therefore, does not include effect plug-ins that introduce a noticeable amount of latency. A further exclusion is the EXS24 mkII Instrument Editor. Effect category Included effects Amp Modeling  Bass Amp (p. 19)  Guitar Amp Pro (p. 21) Delay      Delay Designer (p. 28) Echo (p. 46) Sample Delay (p. 46) Stereo Delay (p. 47) Tape Delay (p.
Effect category Included effects Modulation            Pitch Pitch Shifter II (p. 155) Reverb       Specialized  Exciter (p. 194)  SubBass (p. 197) Utility  Gain (p. 200)  Test Oscillator (p. 202) Chorus (p. 136) Ensemble (p. 136) Flanger (p. 137) Microphaser (p. 138) Modulation Delay (p. 138) Phaser (p. 140) Ringshifter (p. 142) Rotor Cabinet (p. 147) Scanner Vibrato (p. 148) Spreader (p. 149) Tremolo (p. 150) AVerb (p. 160) EnVerb (p. 161) GoldVerb (p.
1 Amp Modeling 1 You can add the sound of a guitar and bass amplifier to your audio recordings and software instruments. Using a method known as component modeling, both the sound and functionality of musical instrument amplifiers, particularly those used with electric guitar and bass, can be emulated as an effect.
Bass Amp Parameters  Model pop-up menu: Choose from among nine different amplifier models. The choices are: Model Description American Basic 1970s-era American bass amp, equipped with eight 10-inch speakers. Well suited for blues and rock recordings. American Deep Based on the American Basic amp, but with strong lower-mid frequency (from 500 Hz on) emphasis. Well suited for reggae and pop recordings.
 Mid Frequency slider: Sets the center frequency of the mid band (between 200 Hz and 3000 Hz).  Output Level slider: Sets the final output level for the Bass Amp. Guitar Amp Pro The Guitar Amp Pro can emulate the sound of a variety famous guitar amplifiers and the cabinets/speakers used with them. You can process guitar signals directly within Logic Pro, allowing you to reproduce the sound of high-quality guitar amp systems. Guitar Amp Pro can also be used for experimental sound design and processing.
 The Microphone Position section is where you set the position of the microphone on the speaker.  The Microphone Type section is where you choose which type of microphone captures the amp’s sound. Amp Section  Amp pop-up menu: Choose the amp model you want to use. The choices are: Model Description UK Combo 30W Neutral sounding amp, well suited for clean or crunchy rhythm parts. UK Top 50W Quite aggressive in the high frequency range, well suited for classical rock sounds.
Speaker type Description US 1x10 open back Not much resonance in the low frequency range. Suitable for use with (blues) harmonicas. US 1x12 open back 1 Open enclosure of an American lead combo with a single 12" speaker. US 1x12 open back 2 Open enclosure of an American clean/crunch combo with a single 12" speaker. US 1x12 open back 3 Open enclosure of another American clean/crunch combo with a single 12" speaker. US broad range Cabinet simulation of a classic electric piano speaker.
 Master knob: Sets the output volume of the amplifier (going to the speaker). Typically, for tube amplifiers, increasing the Master level produces a more compressed and saturated sound, resulting in a more distorted and powerful (louder) signal. High settings can produce an extremely loud output. In Guitar Amp Pro, the Master parameter modifies the sonic character, and the final output level is set using the Output parameter below the FX section. (see below for information).
When you select either button, the graphic speaker display reflects the current setting. Microphone Type Parameters  Condenser button: When selected, emulates the sound of a studio condenser microphone. The sound of condenser microphones is fine, transparent, and well balanced.  Dynamic button: When selected, emulates the sound of a dynamic cardioid microphone. This microphone type sounds brighter and more cutting, compared to the Condenser model.
2 Delay 2 Delay effects store the input signal—and hold it for a short time—before sending it to the effect input or output. Most delays allow you to feed a percentage of the delayed signal back to the input, creating a repeating echo effect. Each subsequent repeat is a little quieter than the previous one. The delay time can often be synchronized to the project tempo by matching the grid resolution of the project, usually in note values or milliseconds.
Delay Designer Delay Designer is a multi-tap delay. Each tap is an independent delay. Unlike simple delay effects that only offer one or two delays (or taps), Delay Designer offers you up to 26 individual taps. In other words, you can think of Delay Designer as 26 separate delay processors—in one effect unit.
 Tap display: This blue “view screen” display features a graphic representation of all taps. You can see, and edit, the parameters of each tap in this area. See “The Tap Display” section of this chapter for a more detailed look.  Tap parameter bar: Offers a numeric overview of the current parameter settings for the selected tap. You can view and edit the parameters of each tap in this area. See “The Tap Parameter Bar” later in this chapter.
 Identification bar: Includes an identification letter for each tap, along with handles that allow you to move the selected tap backwards or forwards in time. The View Buttons The view buttons determine which parameter is represented in the main display.  Cutoff: When clicked, the taps in the main display will show the highpass and lowpass filter cutoff frequencies.  Reso: When clicked, the main display shows the filter resonance value of each tap.
Note: The Autozoom button needs to be turned off for this to work. When you zoom in on a small group of taps, the overview display continues to show all taps. The area shown in the Tap display is indicated by the bright rectangle. To move to different sections of the Tap display: m Click-hold the bright rectangle and drag to the left or right. The zoomed view in the main display will update as you drag.
3 To finish creating taps, click the Last Tap button. This adds the final tap, ending tap recording, and assigning the last tap as the feedback tap (see “The Master Section” for an explanation of the feedback tap). Note: If you do not click the Last Tap button, tap recording automatically stops after ten seconds, or when the 26th tap is created, whichever comes first. To copy taps in the identification bar: m Option-drag a selection of one or more taps to the desired position.
Deleting Taps To delete a tap, simply select it and press the Delete or Backspace key. You can also select a tap in the identification bar and drag it down, below the Tap display. These methods also work when more than one tap is selected. Finally, you can right-click or Control-click on any tap in the Delay Designer interface, and choose the Delete All Taps command from the shortcut menu to delete all taps. Selecting Taps There will always be at least one selected tap.
m Click the downward pointing arrow in the Tap field of the Tap parameter bar, and choose the desired tap letter from the menu. You can select the next or previous tap by clicking the arrow buttons next to the left of the Tap name. To select multiple taps, do one of the following: m Click-drag across the background of the main display to rubber band select multiple taps. m Shift-click specific taps in the Tap display to select multiple non-adjacent taps.
The Tap Parameter Bar The Tap parameter bar shows the current numeric values for every parameter of the selected tap. You can directly edit these parameters in the Tap parameter bar. The parameters shown are: Â Filter On/Off button: Enables or disables the highpass and lowpass filters for the selected tap. Â HP – Cutoff – LP: You can view, and set, the cutoff frequencies (in Hz) for the highpass and lowpass filters here. Â Slope: Determines how steep the highpass and lowpass filters will be.
Editing Taps You can edit taps both graphically, using the main Tap display, and numerically, using the Tap parameter bar. All tap edits are reflected both graphically and numerically. Editing Taps in the Tap Parameter Bar You can edit every parameter in the Tap parameter bar using standard click, or clickdrag techniques. To edit a parameter in the Tap parameter bar: m Click on a button or up/down arrow to enable, disable, or alter a parameter value. m Drag a parameter value up or down to change it.
If you have multiple taps selected, the values of all selected taps will be increased or decreased relative to other taps. You can also set the value of multiple taps by Command-dragging horizontally and vertically across several taps in the Tap display. As you do so, the parameter value changes to match the mouse position as you drag across the taps.
Editing the Filter Cutoff Parameters in the Tap Display While the steps outlined above apply for most graphically editable parameters, the Cutoff and Pan parameters work in a slightly different fashion. In the Cutoff view, each tap actually shows two parameters—highpass and lowpass filter cutoff frequency.
Editing the Pan Parameter in the Tap Display The way that the Pan parameter is represented in the Pan view is entirely dependent on the input channel configuration of Delay Designer. In mono input/stereo output configurations, all taps are initially panned to the center. To edit the pan position, click-drag (vertically) from the center of the tap—in the direction you wish to pan the tap or taps.
By default, the stereo spread is set to 100%. To adjust this, click drag on either side of the dot. As you do so, the width of the line (extending outwards from the dot) changes. Keep an eye on the spread parameter in the Tap parameter bar to view the spread percentage numerically. In surround configurations, the bright line represents the surround angle. See “Working With Delay Designer in Surround” for more information.
Note: The first time you edit a filter or pitch transpose parameter, the respective module will automatically turn on. This saves you the effort of manually turning on the filter or pitch transposition module before editing. Once you manually turn either of these modules off, however, you will need to manually switch it back on.
Activating Sync Mode Sync mode is turned on or off by clicking the Sync button in the Sync section. An orange ring is shown around the Sync button when Sync mode is on, and a grid that matches the chosen Grid parameter value is shown in the identification bar. Once Sync mode is activated, all taps will move towards the closest delay time value on the grid.
Setting the Swing Value The Swing value determines how close to the absolute grid position every second grid increment will be. A Swing setting of 50% means that every grid increment has the same value. Settings below 50% result in every second increment being shorter in time. Settings above 50% result in every second grid increment being longer in time. To adjust the Swing value: m Click-drag up or down in the Swing field to raise or lower the Swing value.
The Master Section The Master section incorporates parameters for two global functions: delay feedback and dry/wet mix. Using Feedback In simple delays, the only way for the delay to repeat is to use feedback. As Delay Designer offers 26 taps, you can use these to create repeats, rather than requiring discreet feedback controls for each tap. Delay Designer’s global feedback parameter allows you to send the output of one tap back through the effect input, to create a self-sustaining rhythm or pattern.
The Mix Sliders Use the Mix sliders to adjust the level of the dry input signal and the (post-processing) wet signal. Working With Delay Designer in Surround Delay Designer is optimally designed for use in surround configurations.
Echo This simple echo effect always synchronizes the delay time to the project tempo, allowing you to quickly create echo effects that run in time with your composition. Echo Parameters  Time: Sets the grid resolution of the delay time in musical note durations—based on the project tempo. “T” values represent triplets, “.” values represent dotted notes.  Repeat: Determines how often the delay effect is repeated.  Color: Sets the harmonic content (color) of the delay signal.
Stereo Delay The Stereo Delay works much like the Tape Delay (see below), but allows you to set the Delay, Feedback, and Mix parameters separately for the left and right channel. The effect also features a Crossfeed knob for each stereo side. It determines the feedback intensity—or the level at which each signal is routed to the opposite stereo side. You can freely use the Stereo Delay on mono tracks or busses, when you want to create independent delays for the two stereo sides.
Tape Delay The Tape Delay simulates the warm sound of vintage tape echo machines, with the convenience of easy delay time synchronization to your project tempo. The Tape Delay is equipped with a highpass and lowpass filter in the feedback loop, making it easy to create authentic dub echo effects, and also includes an LFO for delay time modulation. The LFO produces a triangular wave, with adjustable speed and modulation intensity.
 Smooth: Evens out the LFO and flutter effect.  Dry and Wet: These individually control the amount of original and effect signal. Setting the Feedback When you set the Feedback slider to the lowest possible value, the Tape Delay generates a single echo. If Feedback is turned all the way up, the echoes are repeated ad infinitum. Note: The levels of the original signal and its taps (echo repeats) tend to accumulate, and may cause distortion.
3 Distortion 3 You can use Distortion effects to recreate the sound of analog or digital distortion, and to radically transform your audio. Distortion effects simulate the distortion created by vacuum tubes, transistors, or digital circuits. Vacuum tubes were used in audio amplifiers before the development of digital audio technology, and are still used in musical instrument amps today.
Bitcrusher The Bitcrusher is a low resolution digital distortion effect. You can use it to emulate the sound of early digital audio, create artificial aliasing by dividing the sample rate, or distort signals until they are unrecognizable. Bitcrusher Parameters  Drive slider and field: Sets the amount of gain (in decibels) applied to the input signal.  Resolution slider and field: Sets the bit rate (between 1 and 24 bits).
Clip Distortion Clip Distortion is a nonlinear distortion effect that produces unpredictable spectra. You can use it to simulate warm, overdriven tube sounds, and also to create drastic distortion. Clip Distortion features an unusual combination of serially connected filters. After being amplified by the Drive value, the signal passes through a highpass filter, and is then subjected to nonlinear distortion, as controlled by the Symmetry parameter.
 Input Gain field and slider (extended parameter): Sets the amount of gain applied to the input signal.  Output Gain field and slider (extended parameter): Sets the amount of gain applied to the output signal. Using Clip Distortion If you set the High Shelving Frequency to around 12 kHz, you can use it like the treble control on a mixer channel strip or a stereo hi-fi amplifier. Unlike those types of treble controls, however, you can boost or cut the signal by up to ±30 dB using the Gain parameter.
Distortion II Distortion II emulates the distortion effect section of a Hammond B3 organ. You can use it on musical instruments to recreate this classic effect, or use it creatively when designing new sounds. Distortion II Parameters . Â PreGain dial: Sets the amount of gain applied to the input signal. Â Drive dial: Sets the amount of saturation applied to the signal. Â Tone dial: Sets the frequency at which the signal is filtered.
Overdrive The Overdrive effect emulates the distortion produced by a field effect transistor (FET), which is commonly used in solid-state musical instrument amplifiers and hardware effects devices. When saturated, FETs generate a warmer sounding distortion than bipolar transistors. Overdrive Parameters  Drive slider and field: Sets the amount of saturation of the transistor.  Tone slider and field: Sets the cutoff frequency at which the signal is filtered.
Phase Distortion The Phase Distortion effect is based on a modulated delay line, similar to a chorus or flanger effect (for more information about these effects, see Chapter 9, “Modulation,” on page 135). Unlike these effects, however, in the Phase Distortion effect the delay time is not modulated by a low frequency oscillator (LFO), but rather by a lowpassfiltered version of the input signal itself. This means that the signal modulates its own phase position.
Using the Phase Distortion The input signal only passes the delay line and is not affected by any other process. The Mix parameter blends the effected signal with the original signal. The delay time is modulated by a side chain signal—namely, the input signal. The input signal passes through a resonant lowpass filter, with dedicated Cutoff frequency and Resonance controls. You can listen to the filtered side chain (instead of the Mix signal) by turning on the Monitor button.
4 Dynamics 4 You can use Dynamics effects to control the perceived loudness of your audio, add focus and punch to tracks and projects, and optimize the sound for playback in different situations. The dynamic range of an audio signal is the range between the softest and loudest parts of the signal (technically, between the lowest and the highest amplitude).
Some compressors, called multiband compressors, can divide the incoming signal into different frequency bands, and apply different compression settings to each band. This helps achieve the maximum level without introducing compression artifacts, and is typically used on an overall project mix. Expanders Expanders are similar to compressors, except that they raise, rather than lower, the signal when it exceeds the threshold. Expanders are used to enliven the audio signal.
Adaptive Limiter The Adaptive Limiter is a versatile tool for controlling the perceived loudness of sounds. It works by rounding and smoothing peaks in the signal, producing an effect similar to an analog amplifier being driven hard. Like an amplifier, it can slightly color the sound of the signal. You can use the Adaptive Limiter to achieve maximum gain without clipping (exceeding 0 dBFS).
Extended Parameters  Mode pop-up menu: Choose which type of peak smoothing the Adaptive Limiter uses from the menu. The choices are OptFit, in which limiting follows a linear curve, which allows signal peaks exceeding 0 dB, and NoOver, which avoids distortion artifacts from the output hardware by ensuring that the signal does not exceed 0 dB.  Lookahead field and slider: Adjusts how far ahead the Adaptive Limiter analyzes the file for peaks.
 Release knob and field: Sets the release time (the amount of time it takes for the compressor to stop reducing the signal once the signal falls below the threshold).  Auto button: When selected, the release time dynamically adjusts to the audio material.  Compression curve display: Shows the compression curve created by the Ratio and Knee parameters, with input as the X-axis and output as the Y-axis.
Using the Compressor The following sections provide information on using each of the main Compressor parameters. Threshold and Ratio The most important Compressor parameters are Threshold and Ratio. The Threshold is the level (in decibels) above which the signal is reduced by the amount set as the Ratio. Because the Ratio is a percentage of the overall level, the more the signal exceeds the threshold, the more it is reduced.
Other Parameters Because the Compressor works by reducing levels, the overall volume of its output is typically lower than the input signal. You can adjust the output level using the Gain slider. You can use the Auto Gain parameter to compensate for the reduction in gain produced by compression, referenced to either –12 dB or 0 dB. Auto Gain sets the level of gain (amplification) to a value of T—(T/R), where T = the Threshold and R = the Ratio.
DeEsser Parameters The Detector parameters are on the left side of the DeEsser window, and the Suppressor parameters are on the right. The center section includes the Detector and Suppressor displays and the Smoothing slider. Detector Section  Detector Frequency knob: Sets the frequency range the DeEsser analyzes.  Detector Sensitivity knob: Sets the degree of responsiveness to the input signal. At higher ratios, the Detector reacts more responsively.
Ducker Ducking is a common technique used in radio and television broadcasting: when the DJ/announcer speaks while music is playing, the music level is automatically reduced. When the announcement has finished, the music is automatically raised to its original volume level. The Ducker plug-in provides a simple means of performing this process. It can even reduce the music level before the speaker starts (but this introduces a small amount of latency).
 Release: Controls how quickly the volume returns to the original level. Set to a high value if you want the music mix to slowly fade up after the announcement. Using the Ducker For technical reasons, the Ducker plug-in can only be inserted in output and bus channels. To use the Ducker plug-in: 1 Insert the Ducker plug-in into an audio or bus channel strip. 2 Assign all track outputs that are supposed to “duck” (dynamically lower the volume of the mix) to a bus.
 (Attack) Time knob: Sets the duration from the beginning of the signal considered as the attack.  Display area: Graphically displays the attack and release curves applied to the signal.  (Release) Time knob: Sets the duration of the signal considered as the release.  (Release) Gain slider: Sets the gain on the release phase of the signal. When set to the center (0) position, the signal is unaffected.  Out Level slider: Sets the level of the output signal.
In contrast to a compressor or expander, the Enveloper operates independently of the absolute level of the input signal—provided the Threshold slider is set to the lowest possible value. Expander The Expander is similar to a compressor except that it increases, rather than reduces, the dynamic range above the Threshold level. You can use the Expander to add liveliness and freshness to your audio, specifically by emphasizing the transients of highly compressed signals.
When using the Expander with Auto Gain active, the signal will sound softer even when the peak level remains the same; in other words, the expander decreases loudness. If you dramatically change the dynamics of a signal (by setting higher Threshold and Ratio values), you may find that you need to reduce the output level using the Gain slider to avoid distortion. In most cases, turning on Auto Gain will adjust the signal to the correct level.
The Lookahead parameter allows the Limiter to look forward in the audio so that it can react earlier to peak volumes by adjusting the amount of reduction. Using Lookahead causes latency, but this latency has no perceptible effect when you use the Limiter as a mastering effect, on previously recorded material. Set Lookahead to higher values if you want the limiting effect to take place before the maximum level is reached, creating a smoother transition.
Multipressor Parameters The parameters in the Multipressor window are grouped into three main areas: the upper graphic display section, the lower set of controls for each frequency band, and the output parameters on the right. Graphic Display Section  Graphic display: Each frequency band is represented graphically. The amount of gain change from 0 dB is shown graphically by the blue bars. For active bands, the band number appears in the center of its area.
 Expnd Thrsh fields (short for Expansion Threshold): Sets the expansion threshold for the selected band. Setting the parameter to its minimum value (–50 dB), means that only signals that fall below this level are expanded.  Expnd Ratio fields (short for Expansion Ratio): Sets the expansion ratio for the selected band.  Expnd Reduct fields (short for Expansion Reduction): Sets the amount of downward expansion for the selected band.
Using the Multipressor In the graphic display, the blue bars show the gain change, not only the gain reduction as in a standard compressor. The gain change displayed is a composite value of the compression reduction + expander reduction + auto gain compensation + gain makeup. Compression Parameters The Compression Threshold and Compression Ratio parameters are the key parameters used to control compression.
Noise Gate The Noise Gate is commonly used to suppress unwanted noise that is audible when the audio signal is at a low level. You can use it to remove background noise, crosstalk from other signal sources, and low-level hum, among other uses. The Noise Gate works by allowing signals above the Threshold level to pass unimpeded while reducing signals below the Threshold level. This allows you to remove lower-level parts of the signal, while allowing the intended parts of the audio to pass.
Using the Noise Gate In most situations, setting the Reduction slider to the lowest possible value ensures that sounds below the Threshold are completely suppressed. Setting it to a higher value attenuates low-level sounds but still allows them to pass. You can also set Reduction to a value greater than 0 (zero) to boost the signal by up to 20 dB. This is useful for ducking effects. The three rotary knobs for Attack, Hold, and Release modify the dynamic response of the Noise Gate.
Silver Compressor The Silver Compressor is a simplified version of the Compressor. It has fewer parameters and requires less CPU power. Silver Compressor Parameters  Gain Reduction display: Shows the amount of compression applied as the audio plays.  Threshold slider and field: Sets the threshold for the Compressor (the level above which the signal is reduced.)  Attack knob and field: Sets the attack time (the amount of time it takes for the compressor to react when the signal exceeds the threshold).
Silver Gate The Silver Gate is a simplified version of the Noise Gate. It has fewer parameters and requires less CPU power. Silver Gate Parameters  Lookahead slider and field: Adjusts how far ahead (in milliseconds) the noise gate analyzes the signal.  Threshold slider and field: Sets the level (in decibels) below which the signal is reduced.  Attack knob and field: Sets the amount of time it takes to fully open the gate after the signal exceeds the threshold.
You can also link channels by assigning them to one of three groups. When you adjust the threshold or output parameter for any channel assigned to a group, that parameter is adjusted by the same amount for all channels assigned to the group. Surround Compressor Parameters The Surround Compressor is divided into three sections: The Link section at the top contains a series of menus where you assign each channel to a group.
Main Section  Ratio slider and field: Sets the ratio by which the signal is reduced when it exceeds the threshold.  Knee knob: Adjusts whether the signal is compressed immediately or more gradually at levels close to the threshold.  Attack knob: Sets the amount of time it takes to reach full compression after the signal exceeds the threshold.  Release knob: Sets the amount of time it takes to return to zero compression after the signal falls below the threshold.
5 EQ 5 EQ (short for Equalization) lets you shape the sound of your audio by changing the level of specific frequency bands. EQ is one of the most commonly used audio effects, both for music projects and in post-production work for video. You can use EQ to shape the sound of an audio file, track, or project by adjusting specific frequencies or frequency ranges. Using EQ, you can create both subtle and extreme changes to the sound of your projects.
Multiband EQs Multiband EQs give you control over a set of filters which together cover a large part of the frequency spectrum. On multiband EQs, you can set the frequency, bandwidth, and Q of each band independently. Using a multiband EQ (such as the Channel EQ, Fat EQ, or Linear Phase EQ), you can perform extensive tone-shaping on any audio source. Multiband EQs are equally useful for shaping the sound of an individual track or an overall project mix.
You can use the Channel EQ in many ways: to shape the sound of individual tracks or audio files, or for tone-shaping on an overall project mix. With its Analyzer and graphic controls, it is very easy to observe the audio signal and make adjustments in real time. Channel EQ Parameters On the left side of the Channel EQ window is the Gain control and parameters for the Analyzer, while the central area of the window includes the graphic display and parameters for shaping each EQ band.
The Q parameter of band 1 and band 8 has no effect when the slope is set to 6 dB/ Oct. When the Q parameter of bands 3 through 6 is set to an extremely high value (such as 100), these filters only affect a very narrow frequency band, and can be used as notch filters. Â Link button: Activates Gain-Q coupling, which automatically adjusts the Q (bandwidth) when you raise or lower the gain on any EQ band, to preserve the perceived bandwidth of the bell curve.
In the graphic display, each EQ band appears as a different color. You can graphically adjust the frequency of a band by dragging horizontally in the area of the band. Drag vertically to adjust the amount of gain for the band (For bands 1 and 8, the slope values can only be changed in the parameter area below the graphic display).
DJ EQ The DJ EQ combines high and low shelving filters, each with a fixed frequency, and one parametric EQ for which you can adjust Frequency, Gain, and Q-Factor. A special feature of the DJ EQ is that it allows the gain of the filters to be reduced up to –30 dB. DJ EQ Parameters  High Shelf field and slider: Sets the amount of gain for the high shelving filter.  Frequency field and slider: Sets the center frequency of the parametric EQ.
 Graphic display: Shows the EQ curve of each frequency band. When you adjust each band’s settings using the controls in the Parameter section, the display reflects your changes immediately. Parameter Section Below the graphic display area are controls which both show the settings for each band, and which you can use to adjust each band’s settings.  Frequency fields: Sets the frequency for each band.  Gain knobs: Sets the amount of gain for each band.
Linear Phase EQ The high-quality Linear Phase EQ effect is similar in appearance to the Channel EQ, sharing the same parameters and eight-band layout. However, the Linear Phase EQ uses a different underlying technology, which preserves the phase of the audio signal 100%—even when you apply the wildest EQ curves to the sharpest signal transients! The Linear Phase EQ uses more CPU resources than the Channel EQ, and introduces greater amounts of latency.
Match EQ The Match EQ effect allows you to store the average frequency spectrum of an audio file as a template and apply the template to your project, so that it matches the spectrum of the original file. Using Match EQ you can acoustically match the sound of different songs you plan to include on an album, or impart the particular sound of any source recording to your own projects.
 Format button: Sets whether the Analyzer displays audio channels via separate curves (L&R for stereo, All Cha for surround) or the summed maximum level (LR Max for stereo, Cha Max for surround).  Select buttons: Click one of the buttons to control whether any changes you make to the filter curve created by matching the template with the current material are applied only to the left channel (L), the right channel (R), or both channels (L+R).
 Smoothing slider and field: Sets the amount of smoothing for the filter curve. A value of 0.0 leaves the filter curve unchanged. At all other settings, the filter curve is smoothed at a constant bandwidth, determined by the set value in semitones. For example, a value of 1.0 means that the smoothing uses a bandwidth of one semitone. A value of 4.0 produces a smoothing bandwidth of four semitones (a major third), a value of 12.0 produces a bandwidth of one octave, and so on.
To use the matched EQ on a track: 1 Set the track you want to match as a sidechain input to the Match EQ. 2 Click the “Template Learn” button, play the entire source audio track from start to finish, then click the “Template Learn” button again. 3 Return to the start of your mix, click Current Material Learn, then play your mix (the current material) from start to finish. 4 When done, click Current Material Match (this automatically disengages the Current Material Learn button).
Note: Each time you match two audio signals—either by loading/learning a new spectrum while Match is activated or by activating Match after a new spectrum has been loaded—any existing changes to the filter curve are discarded, and Apply is set to 100%. By default, the Apply slider is set to 100% when you learn the frequency curve of an audio signal. In many cases, you may want to lower it slightly to avoid extreme spectral changes to your mix.
Single Band EQs Following are descriptions of each of the effects found in the Single Band submenu. High Cut and Low Cut Filter As their names suggest, the Low Cut Filter attenuates the frequency range below the selected frequency, while the High Cut Filter attenuates the frequency range above the selected frequency. Each has a single parameter letting you set the cutoff frequency. High Pass and Low Pass Filter The High Pass Filter affects the frequency range below the set frequency.
Silver EQ The Silver EQ, a Legacy effect, includes three bands: a high shelving EQ, parametric EQ, and low shelving EQ. You can adjust the cutoff frequencies for the high and low shelving EQs, and adjust the center frequency, gain, and Q for the parametric EQ. Silver EQ Parameters  High Frequency field and slider: Sets the cutoff frequency for the high shelving EQ.  Frequency field and slider: Sets the center frequency of the parametric EQ.
Frequency Ranges Used With EQ All sounds can be thought of as falling into one of three basic frequency ranges: bass, midrange, or high (or treble). These can each be further divided to include low bass, low and high midrange, and low and high highs. The following table describes some of the sounds that fall into each range: Name Frequency range Description High High 8–20 kHz Includes cymbal sounds and highest harmonics of instruments.
6 Filter 6 In addition to the filters in EQ effects, you can use filters to change the character of your audio in both familiar and unusual ways. The Filter submenu contains a variety of filter-based effects that you can use to creatively modify your audio, including autofilters, filter banks, vocoders, wah-wah effects, and a gate that uses frequency rather than the amplitude (volume) as the criteria for which part of the signal is allowed to pass through.
AutoFilter The AutoFilter is a versatile filter effect with several unique features. You can use it to create classic, analog-style synthesizer effects, or as a tool for creative sound design. The filter cutoff can be dynamically modulated using both a synthesizer-style ADSR envelope and an LFO (low frequency oscillator).
LFO Section  Coarse and Fine Rate knobs and field: Use together to set the frequency of the LFO. Drag the Coarse slider to set the LFO frequency in Hertz, then drag the Fine slider to fine tune the frequency in 1/000s of a Hertz.  Beat Sync button: When selected, the LFO is synchronized to the sequencer’s tempo.  Phase knob: Lets you shift the phase relationship between the LFO and the sequencer when Beat Sync is active.
Using the AutoFilter The following section provides additional information on using the parameters in the AutoFilter window. Filter Parameters The most important parameters are located on the right side of the AutoFilter window. The Filter Cutoff knob determines the point where the filter kicks in. Higher frequencies are attenuated, while lower frequencies are allowed to pass through. The Resonance knob controls how much frequencies around the cutoff frequency are emphasized.
LFO Parameters You set the waveform of the LFO by clicking one of the Waveform buttons. The choices are: descending sawtooth (saw down), ascending sawtooth (saw up), triangle, pulse wave, or random (random values, Sample & Hold). Once you select a waveform, you can shape the curve with the Pulsewidth slider. Use the Coarse and Fine Frequency knobs to set the LFO frequency. The Rate Mod. (Rate Modulation) knob controls modulation of the LFO frequency independent of the input signal level.
EVOC 20 Filterbank Parameters The EVOC 20 Filterbank window is divided into three main sections: the Formant Filter section in the center of the window, the Modulation section at the bottom center, and Output section along the right side. Formant Filter Section The parameters in this section control the frequency bands in the two filter banks: Filter Bank A and Filter Bank B.
 Formant Shift knob: Moves the position of all bands in both filter banks up or down the frequency range. You can jump directly to the values –0.5, –1, 0, +0.5 or +1.0 by clicking one of these numbers on the edge of the knob.  Bands value field: Sets the number of frequency bands in each filter bank. The range is from 5 to 20 bands. Note: Increasing the number of bands also increases the CPU overhead.  Lowest button: Sets whether the lowest band of each filter bank acts as a lowpass or bandpass filter.
 LFO Shift Intensity slider: Sets the amount by which the LFO modulates the Formant Shift parameter.  LFO Shift Rate knob: Sets the speed of the modulation for Formant Shift. Values to the left of center are synchronized to the tempo in bars and other musical values, while values to the right of center are free values in Hertz.  Waveform buttons: Select the waveforms used by the LFO Shift and LFO Fade LFOs respectively.
The parameters in this section control the overall output of the EVOC 20 Filterbank. Â Overdrive button: Turns the overdrive circuit on or off. Note: To hear the Overdrive effect, you may need to boost the level of one or both filter banks. Â Level slider: Sets the level of the output signal. Stereo Mode pop-up menu: Sets the input/output mode of the EVOC 20 Filterbank. The choices are m/s (mono input to stereo output), and s/s (stereo input to stereo output).
∏ Tip: For good pitch tracking, it is essential to use a mono signal (with no overlap of pitches) that is as unprocessed as possible. Avoid using a signal with background noises. Using a signal processed with even a slight amount of reverb, for example, will produce strange (and likely undesirable) results. Even stranger results will result when a signal with no audible pitch (such as drum loop) is used. In some situations, however, the resulting artifacts might be desirable.
Note: A long attack time on percussive input signals (such as spoken word or hi-hat parts) results in a less articulated vocoder effect. Set Attack as low as possible to achieve precise articulation. Â Release knob: Controls how quickly the envelope follower coupled to each analysis filter band reacts to falling signals. Longer release times make transients of the analysis input signal sound for a longer period of time at the Vocoder’s output.
 Oscillator (Osc.): Sets the tracking oscillator as the synthesis source. The oscillator tracks the pitch of the analysis input signal. Choosing Osc. activates the other parameters in the Synthesis section. If Osc is not chosen, the FM Ratio, FM Int, and other parameters in this section have no effect.  Track: Sets the audio track into which the EVOC 20 TrackOscillator is inserted as the synthesis source signal.
 An FM Ratio of 3.000 produces results resembling a square wave with a pulse width of 33%.  FM Int knob: Selects the basic waveform and controls the intensity of FM modulation.  At a value of 0, the FM tone generator is disabled, and a sawtooth wave is generated instead.  For values higher than 0, the FM tone generator is activated. Higher values result in a more complex and brighter sound.  Coarse Tune value field: Sets the pitch offset of the oscillator in semitones, up to ±2 octaves.
 You can add notes to the chosen scale or chord by clicking keys on the small keyboard, and remove notes by clicking notes already selected. Selected notes appear bright green. Selecting any notes sets the Root/Scale value to user.  Your last edit is remembered. If you choose a new scale or chord, but do not make any changes, you can jump back to the previously set user scale. You can automate the Root and Scale parameters and the keys of the onscreen keyboard in Logic Pro.
 Highest button: Sets whether the lowest band of each filter bank acts as a highpass or bandpass filter.  Formant Stretch knob: Alters the width and distribution of the bands in the synthesis filter bank, extending or narrowing the frequency range defined by the blue bar (Low/High frequency parameters) for the synthesis filter bank. If Formant Stretch is set to 0, the width and distribution of the bands in the synthesis filter bank is equal to the width of the bands in the analysis filter bank.
The parameters in this section control the LFO that can be used to modulate either the frequency (Pitch) of the tracking oscillator (vibrato), or the Formant Shift (Shift) parameter of the synthesis filter bank. It allows synchronous/non-synchronous modulation in bar, beat (triplet) or free values. Â Wave buttons: Select the waveform used by the LFO.
 Sensitivity knob: Sets the degree of responsiveness of U/V detection. By turning this knob to the right, more of the individual unvoiced portions of the input signal are recognized. When high settings are used, the increased sensitivity to unvoiced signals can lead to the U/V source—determined by the Mode parameter—being used on the majority of the input signal, including voiced signals.
 Signal pop-up menu: Choose the signal to send to the EVOC 20 TrackOscillator’s main outputs. The choices are: Voc(oder), Syn(thesis), and Ana(lysis). To hear the vocoder effect, choose Voc. The other two settings are useful for monitoring purposes.  Level slider: Sets the level of the output signal.  Stereo Mode pop-up menu: Sets the input/output mode of the EVOC 20 Filterbank. The choices are m/s (mono input to stereo output), and s/s (stereo input to stereo output).
Wah Section  Wah Mode pop-up menu: Choose one of the six modes, which emulate various classic wah effects and filter types, or choose off.  Auto Gain button: The wah effect can cause the output level to vary widely. Turning Auto Gain On compensates for this tendency, and keeps the output signal within a more stable range.  Wah Level knob: Sets the amount of the wah-filtered signal.  Relative Q slider: Adjusts the sharpness of the wah sweep by raising or lowering the filter peak.
3 Make a sweep with a high relative Q setting. 4 Switch Auto Gain to off, and repeat the sweep. Warning: Please take care while doing this, or your ears and speaker system may be damaged. AutoWah Depth In addition to using MIDI foot pedals (see above), the wah effect can be controlled using the Auto Wah facility. The sensitivity of the Auto Wah can be set with the Depth parameter. Range: 0.00 to 100. (See also “Envelope (Depth)” on page 374.
 Threshold slider and field: Sets the threshold level at which the frequency band defined by the Center Freq. and Bandwidth parameters is divided into upper and lower frequency ranges.  Speed slider and field: Sets the modulation frequency for the defined frequency band.  CF (Center Frequency) Modulation slider and field: Sets the intensity of center frequency modulation.  BW (Band Width) Modulation slider and field: Sets the amount of bandwidth modulation.
You can modulate the defined frequency band using the Speed, CF Modulation, and BW Modulation parameters. Speed determines the modulation frequency, CF (Center Frequency) Modulation defines the intensity of the center frequency modulation, and BW (Band Width) Modulation controls the bandwidth modulation. After making your adjustments, you can use the Gain slider to adjust the final output level of the processed signal.
7 Imaging 7 You can use the Logic Studio Imaging plug-ins to extend the stereo base of a recording, and to alter perceived signal positions. These effects enable you to make certain sounds, or the overall mix, seem wider and more spacious. You can also alter the phase of individual sounds within a mix, to enhance or suppress particular transients. The following sections describe the Imaging plug-ins included with Logic Studio: Â “Binaural Post-Processing Plug-in” on page 121.
The Binaural Post-Processing plug-in is available in aux and output channels. The Binaural Post-Processing plug-in offers the following Compensation choices: Â Headphone FF - optimized for front direction: Setting for headphone playback, utilizing free-field compensation. In this compensation mode, sound sources placed in front of the listening position will have neutral sound characteristics. Â Headphone HB - optimized for horizontal directions: Setting for headphone playback.
Direction Mixer You can use the Direction Mixer plug-in to decode middle and side (MS) audio recordings (see “What Is MS?” on page 124), or to spread the stereo base of a (left/right) recording, and determine its pan position. Â Input buttons: Use the LR or MS buttons to determine whether the input signal is a standard left/right signal, or if you’re dealing with an MS encoded (middle and side) signal. Â Spread slider and field: Determines the spread of the stereo base.
Setting the Direction Parameter When Direction is set to a value of 0, the middle of the stereo recording will be dead center within the mix. If you use positive values, the midpoint of the stereo recording is moved towards the left. Negative values move the midpoint to the right. Here’s how this works: Â At 90˚, the midpoint of the stereo recording is panned hard left. Â At –90˚, the midpoint of the stereo recording is panned hard right.
Here’s some interesting trivia for you: Radio (FM) broadcasts feature M and S stereo. The MS signal is actually converted to a signal suitable for the left and right speakers by the receiver. Stereo Spread The Stereo Spread effect is typically used for mastering. There are several ways to extend the stereo base (or perception of space), including use of reverbs and other effects, and altering the signal’s phase.
 Upper and Lower Freq. slider and fields: Use these to determine the upper and lower limits of the highest frequency, and lowest band, to be distributed in the stereo image.  Order knob: Sets the number of frequency bands that the signal is divided into. A value of 8n is usually sufficient for most tasks, but you can use up to 12 bands.
8 Metering 8 You can use the Metering plug-ins of Logic Studio to analyze audio in a variety of ways. Each Metering plug-in allows you to view different characteristics of an audio signal. As examples: The BPM Counter displays the tempo of an audio file, the Correlation Meter displays the phase relationship, and the Level Meter displays the level of an audio recording.
BPM Counter You can use the BPM Counter to analyze the tempo of an audio track. Insert the plug-in into a track, to analyze the dynamic events of the audio signal. The detection circuit looks for any transients in the input signal. Transients are very fast, non-periodic sound events in the attack portion of the signal. The more obvious this impulse is, the easier it is for the BPM Counter to detect the tempo.
Level Meter The Level Meter displays the current signal level on a decibel scale. The signal level for each channel is represented by a blue bar. When the level exceeds 0 dB, the portion of the bar above the 0 dB point becomes red. Stereo instances of the Level Meter show independent left and right bars, while mono instances display only a single bar. Surround instances display a bar for each channel, in a vertical rather than horizontal orientation.
MultiMeter Parameters You can view either the Analyzer or Goniometer in the main display area. You switch the view and set other MultiMeter parameters using the controls on the left side of the window. The Phase Correlation meter is always visible at the bottom of the window, and the Level Meters are visible on the right. Analyzer Section  Analyzer button: When selected, displays the Analyzer in the center of the window.
 In the Correlation Meter, a growing horizontal area around the white correlation indicator shows any phase correlation deviations—in both directions. A vertical red line to the left of the correlation indicator permanently shows the maximum negative phase deviation value. You can reset this line by clicking on it during playback.  In the Level Meter, a small yellow segment above each stereo level bar labels the most recent peak level.
Goniometer A Goniometer helps you to judge the coherence of the stereo image and determine phase differences between the left and right channels. Phase problems are easily spotted as trace cancelations along the center line (M—mid/mono). The idea of the Goniometer was born with the advent of early two-channel oscilloscopes.
Surround MultiMeter The surround version of the MultiMeter includes some additional parameters for use in analyzing multichannel surround files. Surround MultiMeter Parameters Analyzer Section These parameters are similar to those in the stereo MultiMeter, but include buttons for each surround channel. You can select a single channel or a combination of channels.
Tuner You can tune both acoustic and electric music instruments connected to your system using the Tuner. Tuning your instruments ensures that your recordings will be in tune with any software instruments, existing samples, or existing recordings in your projects. Tuner Parameters  Graphic tuning display: As you play, the pitch of the note appears in the semicircular area, centered around the Keynote.
9 Modulation 9 Modulation effects are used to add motion and depth to your sound. Modulation effects include chorus, flanging, and phasing among others, which make sounds richer or more animated. This is often achieved through the use of an LFO, which is controlled with parameters such as speed or frequency, and depth (also called width, amount, or intensity). You can also control the ratio of the affected (wet) signal and the original (dry) signal.
Chorus The Chorus effect delays the original signal. The delay time is modulated with an LFO. The delayed, modulated signal is mixed with the original, dry signal. You can use the Chorus effect to enrich the sound and create the impression that it’s being played by multiple instruments or voices, in unison. The slight delay time variations generated by the LFO simulate the subtle pitch and timing differences heard when several people perform together.
 Phase knob and field: Controls the phase relationship between the individual voice modulations. The value that you choose here is dependent on the number of voices, which is why it is shown as a percentage value rather than degrees. The value 100 (or –100) is equal to the greatest possible distance between the modulation phases of all voices.  Spread slider and field: Used to distribute the voices across the stereo or surround field.
Microphaser The Microphaser is a simple phaser effect that allows you to quickly create swooshing, phasing effects with just three parameters: Â LFO Rate slider and field: Defines the frequency, and therefore the speed, of the LFO. Â Feedback slider and field: Determines the amount of the effect signal that is routed back into the input. Â Intensity slider and field: Determines the amount of modulation.
 LFO 1 and LFO 2 Rate knobs and fields: Use the left knob to set the modulation rate for the left stereo channel, and the right knob to set the modulation rate of the right stereo channel. The right LFO Rate knob is only available in stereo and surround instances, and can only be set separately if the Left Right Link button is not enabled. In Surround instances, the center channel is assigned the middle value of the LFO left and LFO right Rate knobs.
Phaser The Phaser effect combines the original signal with a copy that is slightly out of phase with the original. This means that the amplitude of the two signals reach their highest and lowest points at slightly different times. The timing differences between the two signals is modulated by two independent LFOs. In addition, the Phaser includes a filter circuit and a built-in envelope follower, which tracks any volume changes in the input signal, generating a dynamic control signal.
 Distribution menu (only available in surround instances): Defines how the phase offsets between the individual channels are distributed in the surround field. You can choose from Circular, Random, Front <> Rear, and Left <> Right distribution. When you load a setting that uses the Random option, the saved phase offset value is recalled. If you want to randomize the phase setting again, choose “new random“ in the Distribution menu.
Ringshifter The Ringshifter effect combines a ring modulator with a frequency shifter effect. Both effects were popular during the 1970s, and are currently experiencing something of a renaissance. Â The ring modulator modulates the amplitude of the input signal using either the internal oscillator or a side chain signal. The frequency spectrum of the resulting effect signal equals the sum and difference of the frequency content in the two original signals.
Modes The four mode buttons determine whether the Ringshifter operates as a frequency shifter or as a ring modulator. Â Single (Frequency Shifter) button: The frequency shifter generates a single, shifted effect signal. The oscillator Frequency control determines whether the signal is shifted up (positive value) or down (negative value). Â Dual (Frequency Shifter) button: The frequency shifting process produces one shifted effect signal for each stereo channel—one is shifted up, the other is shifted down.
 Frequency control: Sets the frequency of the sine oscillator.  Lin(ear) and Exp(onential) buttons: Use these buttons to switch the scaling of the Frequency control:  The exponential scaling offers extremely small increments around the 0 point, which is useful for programming slow moving phasing and tremolo effects.  In the Lin(ear) mode, the resolution of the scale is even across the entire control range.
Output  Feedback knob and field: Sets the amount of the signal that is routed back to the effect input.  Stereo Width knob and field: Determines the breadth of the effect signal in the stereo field. Stereo Width only affects the effect signal of the Ringshifter, not the dry input signal.  Dry/Wet knob and field: Set the mix ratio of the dry input signal and the wet effect signal.
Modulation Sources The oscillator Frequency and Dry/Wet parameters can be modulated via the internal envelope follower and LFO. The oscillator frequency even allows modulation through the 0 Hz point, thus changing the oscillation direction. Envelope Follower The envelope follower analyzes the amplitude (volume) of the input signal and uses this to create a continuously changing control signal—a dynamic volume envelope of the input signal. This control signal can be used for modulation purposes.
Rotor Cabinet The Rotor Cabinet effect emulates the rotating loudspeaker cabinet of a Hammond organ’s Leslie effect. It simulates both the rotating speaker cabinet, with and without deflectors, and the microphones which pick up the sound. Â Rotor speed buttons: These switch the rotor speed. Chorale switches to slow movement, Tremolo to fast movement, and Brake stops the rotor. Â Cabinet Type menu: Choose between various cabinet sizes, shapes, and material.
 Wood & Horn IR: This setting uses an impulse response (a recording) of a Leslie with a wooden enclosure.  Proline & Horn IR: This setting uses an impulse response of a Leslie with a more open enclosure.  Split & Horn IR: This setting uses an impulse response of a Leslie with the bass rotor signal routed more to the left side, and the treble rotor signal routed more to the right side.
 Stereo Phase knob: If set to a value between 0 and 360 degrees, Stereo Phase determines the phase relationship between left and right channel modulations, thus enabling synchronized stereo effects. If “free” is chosen, you can set the modulation speed of the left and right channel independently.  Rate Left knob: Sets the modulation speed of the left channel when Stereo Phase is set to free.
Tremolo The Tremolo effect modulates the amplitude of a signal, resulting in periodic volume changes. You’ll recognize this effect from vintage guitar combo amps (where it is sometimes incorrectly referred to as vibrato). The graphic display shows all parameters, except Rate. Â Depth slider and field: Determines the modulation amount. Â Rate knob and field: Defines the frequency, and therefore the speed, of the LFO. Â Symmetry and Smoothing knobs and fields: Use these to set the shape of the modulation.
10 Pitch 10 You can use the Pitch effects of Logic Studio to transpose or correct the pitch of audio tracks. These effects can also be used for creating unison or slightly thickened parts, or even the creation of harmony voices. Logic Studio includes the following Pitch effects: Â “Pitch Correction” on page 151 Â “Pitch Shifter II” on page 155 Â “Vocal Transformer” on page 156 Pitch Correction You can use the Pitch Correction plug-in to correct the pitch of audio tracks.
Pitch Correction Parameters  Normal and Low buttons: These determine the pitch range that is scanned (for notes that need correction).  Use Global Tuning button: Enable to use the project’s Tuning settings for the pitch correction process. If this button is switched off, you can use the Ref. Pitch field to freely set the desired reference tuning, in cents.  Scale field: Click to choose different pitch quantization grids from the Scale menu.  Root field: Click to choose the root note of the scale.
Defining the Pitch Quantization Grid The Scale menu allows you to choose different pitch quantization grids. The scale that is set manually (with the keyboard) is called the User Scale. The default setting is the chromatic scale. The other scale names are self-explanatory. If you’re unsure of the intervals used in any given scale, simply choose it in the Scale menu and check out the values shown on the keyboard. You can alter any note in the scale by clicking on the keyboard keys.
∏ Tip: You’ll often find that it’s best to only correct notes with the most harmonic gravity. As an example, choose the “sus 4” Scale, and set the Root note to match the project key. This will limit correction to the root note, the fourth and the fifth of the key scale. Switch all other notes to Bypass, and only the most important and sensitive notes will be corrected, while all other singing remains untouched.
If you keep a close eye on this display, you can use it for two important tasks: To better understand the inner workings of the algorithm, and adjust the Response accordingly. You can also use the display when discussing (and optimizing) the vocal intonation with a singer during a recording session. Automating the Pitch Correction Plug-in Pitch Correction can be fully automated. This means that you can automate the Scale and Root parameters to follow the harmonies of the song.
 Timing pop-up menu (extended parameters): Sets whether the timing follows the selected preset (Preset), creates a new preset by analyzing the incoming signal (auto), or uses the settings of the Delay, Crossfade, and Stereo Link parameters, described below (manual). The following three parameters are active only when manual is chosen in the Timing pop-up menu.  Delay slider and field: Sets the amount of delay applied to the input signal.
The Vocal Transformer is well suited to extreme vocal effects. The best results are achieved with monophonic signals, including monophonic instrument tracks. The plugin is not designed for polyphonic voices (a choir on a single track, for example) or other chordal tracks. Vocal Transformer Parameters  Pitch knob and field: Determines the amount of transposition applied to the input signal.
Setting the Pitch and Formant Parameters The Pitch parameter transposes the pitch of the signal up to two octaves upwards or downwards. Adjustments are made in semitone steps. Incoming pitches are indicated by a vertical line below the Pitch Base field. Transpositions of a fifth upward (Pitch = +7), a fourth downward (Pitch = –5), or by an octave (Pitch = ±12) are the most useful, harmonically. As you alter the Pitch parameter, you might notice that the formants don’t change.
11 11 Reverb You can use Reverb effects to simulate the sound of acoustic environments such as rooms, concert halls, caverns, or the sound of infinite space. Sounds bounce off the surfaces of any space, or off objects within a space, repeatedly, gradually dying out until they are inaudible. The bouncing soundwaves result in a reflection pattern, more commonly known as a reverberation (or reverb).
Digital recording introduced digital reverb effects, which consist of thousands of delays of varying lengths and intensities. The time between the original signal and the arrival of the early reflections can be adjusted by a parameter commonly known as predelay. The average number of reflections in a given period of time is determined by the density parameter. The regularity or irregularity of the density is controlled with the diffusion parameter.
 Reflectivity: Defines how reflective the imaginary walls, ceiling, and floor are. How hard the walls are, and what they’re made of, in other words; glass, stone, timber, carpet, and other materials have a dramatic impact on the tone of the reverb.  Room Size: Defines the dimensions of simulated rooms.  Density/Time: Determines both the density and duration of the reverb.  Mix: Determines the balance between the effected (wet) and direct (dry) signals.
 Hold: Sets the duration (time) of the sustain phase.  Release: Sets the time that the reverb takes to fade out completely, after it has completed the sustain phase. Sound Parameters  Density: Sets the reverb density.  Spread: Controls the stereo image of the reverb. At 0%, the effect generates a monaural reverb. At 200%, the stereo base is artificially expanded.  High Cut: Frequencies above the set value are filtered out of the reverb tail.
Early Reflection Parameters  Predelay: Determines the amount of time between the start of the original signal, and the arrival of the early reflections.  Room Shape: Defines the geometric form of the room. The numeric value (3 to 7) represents the number of corners in the room. The graphic display visually represents this setting.  Room Size: Determines the dimensions of the room. The numeric value indicates the length of its walls—the distance between two corners.
Setting Density and Diffusion Ordinarily, you want the signal to be as dense as possible. However, use of a lower Density value means the effect eats up less computing power. Beyond this, in rare instances, a high Density value can color the sound, which you can fix by simply reducing the Density knob value. Conversely, if you select a Density value that is too low, the reverb tail will sound grainy.
The interface can be broken down into four parameter groups: Â Early Reflections parameters: Emulates the original signal’s first reflections as they bounce off the walls, ceiling, and floor of a natural room. Â Reverb parameters: Controls the diffuse reverberations. Â Balance ER/Reverb parameter: Controls the balance between the Early Reflections and Reverb section. When you set the slider to either of its extreme positions, the unused section is deactivated.
Output Parameters  Dry: Controls the amount of the original signal.  Wet: Controls the amount of the effect signal. Setting Predelay and Initial Delay In practice, too short a Predelay tends to make it difficult to pinpoint the position of the signal. It can also color the sound of the original signal. On the other hand, too long a Predelay can be perceived as an unnatural echo. It can also divorce the original signal from its early reflections, which leaves an audible gap.
Setting the Reverb Time and Level of the Low Frequency Band You can use the Low Ratio control to offset the reverb time of the low frequency band. At 100%, the reverb times for the two bands are identical. At lower values, the reverb time of the frequencies below the crossover frequency is shorter. At values greater than 100%, the reverb time for low frequencies is longer. Both of these phenomena occur in nature. In most mixes, a shorter reverb time for bass frequencies is preferable.
 Modulation Phase: Defines the phase of the modulation between the left and right channels of the reverb signal. At 0°, the extreme values (minimum or maximum) of the modulation are achieved simultaneously on both the left and right channels. At a value of 180°, the extreme values opposite each other (left channel minimum, right channel maximum, or vice-versa) are reached simultaneously.  Mod. Intensity: Sets the modulation amount. A value of 0 turns the delay modulation off.
12 Convolution Reverb: Space Designer 12 Space Designer is a convolution reverb effect. You can use it to create exceptionally realistic reverberations. Space Designer generates reverb by convolving, or combining, an audio signal with an impulse response (IR) reverb sample. For example, imagine that you apply the Space Designer to a vocal track.
You can use Space Designer to create both a highly realistic reverb if you use an IR recorded from a real space, or a completely unique effect if you use a synthesized IR that doesn’t represent any real space. Convolution can be used to put your audio signal inside any space, including a speaker cabinet, plastic toy, and so on, if you make an IR from it. And with Space Designer’s extensive audio processing features, you can exactly tailor its space to your material.
 Filter parameters: You can further modify the timbre of the Space Designer reverb using these resonant filter parameters. You can choose from a number of filter modes, adjust the resonance of the filter, as well as adjust the filter envelope dynamically over time, as you can with the volume envelope. See “Filter Parameters” on page 184.  Synthesized impulse response parameters: If you do enough processing of the original IR, you may want to synthesize a new IR from your edited parameters.
 IR Sample Arrow button: Click to load an impulse response.  Sample Rate parameter: Determines the sample rate of the loaded impulse response.  Preserve Length option: Activate to preserve the length of the impulse response when changing the sample rate.  Length parameter: Adjusts the length of the impulse response.  Synthesized IR button: Click to switch Space Designer to Synthesized IR mode.
Setting the Sample Rate The Sample Rate slider is used to determine the sample rate of an impulse response. You can choose between the following settings: Â Orig: Space Designer uses the current project sample rate. When loading an impulse response, Space Designer automatically converts the sample rate of the impulse response to match the current project sample rate—should it be necessary. As an example, this allows you to load a 44.1 kHz impulse response into a project running at 96 kHz, and vice versa.
Similar adjustments can be made while running in Synthesized IR mode. Most typical reverb sounds don’t feature an excessive amount of high frequency content. If you were running at 96 kHz, you would need to make use of some deep lowpass filtering to obtain the mellow frequency response characteristics of many reverb sounds. As a different approach, you are better served to first reduce the high frequencies by 1/2 or even 1/4 using the Sample Rate slider, and then apply the lowpass filter.
 Latency Compensation button: Switches Space Designer’s internal latency compensation feature on or off. See “Latency Compensation” on page 177 for more information.  Definition area: Lets you configure Space Designer to switch to a less defined IR set in order to emulate reverb diffusion and save CPU. See “Definition” on page 177 for more information.  Rev Vol Compensation: Engages Space Designer’s internal IR volume matching (see “Rev Vol Compensation” on page 177 for more information).
Input Slider The Input slider functions as either a stereo processing slider for stereo instances of Space Designer, or as an LFE to Reverb slider in surround mode. The Input slider does not appear in mono or mono to stereo instances of Space Designer. Stereo Mode For stereo instances of Space Designer, the Input slider determines how a stereo signal is processed: Â Stereo setting (top of slider): The signal is processed on both channels, retaining the stereo balance of the original signal.
Latency Compensation The complex calculations made by Space Designer take time. This time results in a processing latency, or delay, between the direct (input) signal, and the processed (output) signal. The Latency Compensation button determines how Space Designer delays the direct signal in relation to the processed signal. Space Designer’s processing latency is 128 samples at 44.1 kHz, and doubles at each lower sample rate division.
Natural reverbs contain most of their spacial information in the first few milliseconds. Towards the end of the reverb, its reflection pattern diffuses more and more, containing less spacial information. In order to emulate this phenomenon—as well as conserve CPU power—you can configure Space Designer to only use the full IR resolution at the onset of the reverb, and to use a reduced IR resolution towards the end of the reverb.
Surround Configuration In surround configurations, Space Designer offers four output sliders that together comprise a small surround output mixer. These sliders have the following functions:  C(enter): Adjusts the center reverb level  Bal(ance): Sets the balance between the L–C–R front and the Ls–Rs rear speakers. In 7.1 ITU surround, the balance pivots around the Lm–Rm speakers, taking the surround angles into account. With 7.1 SDDS surround, the Lc–Rc speakers are considered front speakers.
IR Start The IR Start parameter enables you to shift the playback point into the impulse response, which will effectively cut off the beginning of the impulse response. The IR Start parameter can for example be used to eliminate any peaks at the beginning of the impulse response sample. It also offers a number of creative options, such as its use when combined with the Reverse function (see “Button Bar” on page 181). Note: The IR Start parameter is not available in Synthesized IR mode.
Button Bar The Envelope and EQ display’s button bar includes buttons to switch the main display between envelopes and the EQ, as well as some function buttons. Â Reset button: Click to reset the currently displayed envelope or EQ to its default values. Â All button: Click to reset all envelopes and the EQ to default values. Â Volume Env button: Click this button to bring the volume envelope to the front of the main display. The other envelope curves are shown as transparencies in the background.
 Zoom to Fit button: Switch on to display the entire impulse response waveform. The display will automatically follow any envelope length changes.  A and D buttons: Click to limit the Zoom to Fit function to the attack and decay portions of the (currently selected) envelope. The A and D buttons are available only to the volume and filter envelopes. Setting Envelope Parameters Space Designer allows you to edit the volume and filter envelopes of all IRs, and the density envelope of synthesized IRs.
Volume Envelope Parameters The volume envelope lets you set the reverb’s initial level and adjust how the volume will change over time. You can edit all of the volume envelope parameters numerically, and many of them can also be edited graphically using the techniques discussed in “Setting Envelope Parameters” on page 182.
Filter Parameters Space Designer’s filter provides control over the timbre of the reverb. Its controls are distributed between two parts of the Space Designer interface: The main filter parameters are found in Space Designer’s lower left corner, and the filter envelope appears on the Envelope and EQ display when its Filter button is engaged. You can select from several filter types, but you also have envelope control over the filter cutoff, independent from the volume envelope.
Filter Envelope Parameters The filter envelope lets you control the filter’s cutoff frequency over time. All of the filter envelope’s parameters can be adjusted either numerically in the parameter area or graphically in the main display using the techniques discussed in “Setting Envelope Parameters” on page 182. Controls the Attack Time endpoint (and Decay Time startpoint) and Break Level parameters simultaneously. Controls the Decay endpoint and End Level parameters simultaneously.
Synthesizer Impulse Response Parameters In Synthesizer IR mode, Space Designer generates a synthesized impulse response determined by the values of the Length, envelopes, filter, EQ, and spread parameters. To switch to Synthesizer IR mode, enable the Synthesized IR button in the impulse response parameters section. Clicking the activated Synthesized IR button will randomly generate new impulse responses with slightly different reflection patterns.
 End Level: Sets the density of the reverb tail. If you select an End Level value that is too low, the reverb tail will sound grainy. You may also find that the stereo spectrum is affected by lower values.  Reflection Shape: Determines the steepness (shape) of the early reflection clusters as they bounce off the walls, ceiling, and furnishings of the virtual space. Small values result in clusters with a sharp contour, and large values result in an exponential slope and a smoother sound.
EQ Parameters Space Designer features a four-band EQ comprised of two parametric mid-bands plus two shelving filters (one low shelving filter and one high shelving filter). The EQ has the following parameters: Â EQ On/Off button: Click to switch the entire EQ section on or off. Â Individual EQ buttons (1 through 4): Click to turn individual EQ bands on or off. Â Frequency: Sets the frequency for the selected EQ band. Â Gain: Adjusts the gain cut or boost for the selected EQ band.
 Drag any band to the right or left to adjust its frequency.  Drag any band up to increase the Gain, down to lower the Gain.  Position the mouse pointer directly on the (illuminated) pivot point of a parametric band and drag up to raise the Q or down to lower the Q. Automating Space Designer Space Designer cannot be fully automated as per most other Logic Studio plug-ins.
13 Specialized 13 Logic Studio includes a bundle of specialized plug-ins designed to address tasks often encountered during audio production. You should have a look at these specialized effects if you want to do one of the following: Â Eliminate or reduce noise below a threshold level (see “Denoiser” on page 192). Â Enhance the timing of audio recordings (see “Enhance Timing” on page 193). Â Add life to digital recordings by adding additional high frequency components (see “Exciter” on page 194).
Denoiser The Denoiser eliminates or reduces any noise below a threshold volume level. Denoiser Parameters  Threshold slider and field: Sets the volume level (the threshold) below which the DeNoiser reduces the signal.  Reduce slider and field: Sets the amount of noise reduction applied to sounds below the threshold. When reducing noise, remember that each 6 dB reduction is equivalent to halving the volume level (and each 6 dB increase equals a doubling of the volume level).
 Graphic display: Shows how the lowest volume levels of your audio material (which should be mostly or entirely noise) are reduced. Changes to parameters are instantly reflected here, so keep an eye on it! Using the Denoiser Locate a section of the audio where only noise is audible, and set the Threshold value so that only signals at, or below, this level are filtered out.
Using the Enhance Timing Effect The Enhance Timing plug-in is designed to tighten up loose playing (of recorded audio) in a production. It can be used on a variety of material, and works in real time. Obviously, this type of real-time quantization has some limitations. It will not work well on recordings of performances that have been played too far off the beat. The same is true for very complex, layered drum tracks.
 Input button: When selected, the original (pre-effect) signal is mixed with the effected signal. If you disable Input, only the effected signal is heard.  Harmonics knob and field: Sets the amount of the effected signal that is mixed with the original signal (expressed as a percentage). If the Input button is turned off, this has no effect on the signal.
 Grid buttons: Determine the beat division used as a timing reference by the algorithm to analyze the audio material. Choose 1/8th if the audio material contains primarily eighth notes, and 1/16 if it consists mostly of sixteenth notes.  Accent slider and field: Raises or lowers the level of even beats, accentuating them. Such accents are typical of a variety of rhythmic styles, such as swing or reggae.
SubBass The SubBass plug-in generates frequencies below those of the original signal—in other words, an artificial bass. The simplest use for the SubBass is as an octave divider, similar to Octaver effect pedals for electric bass guitars. Where such pedals can only process a monophonic input sound source of clearly defined pitch, SubBass can be used with complex summed signals as well. SubBass creates two bass signals, derived from two separate portions of the incoming signal.
 Dry slider and field: Sets the amount of dry (non-effected) signal.  Wet slider and field: Sets the amount of wet (effected) signal. Using the SubBass Unlike a pitch shifter, the waveform of the signal generated by the SubBass is not based on the waveform of the input signal, but is sinusoidal (it uses a sine wave). Given that pure sine waves rarely sit well in complex arrangements, you can control the amount of (and balance between) the generated and original signals using the Dry and Wet sliders.
14 Utility 14 The Utility plug-ins are handy tools that can help you with routine tasks and situations that you may encounter when producing music. This includes the following tasks: Â Adjusting the input format of a channel (see “Down Mixer” on page 199). Â Adjusting the level or phase of input signals (see “Gain” on page 200 and “Multichannel Gain” on page 202). Â Integrating external audio effects into Logic Pro (see “I/O” on page 201).
Gain Gain lets you amplify (or reduce) the signal by a specific decibel amount. It is very useful when you are working with automated tracks during post-processing and want to quickly adjust levels. As examples: when you have inserted another effect that doesn’t have its own gain control, or when you want to change the level of a track for a remix version. Gain Parameters  Gain slider and field: Sets the amount of gain.
I/O The I/O plug-in allows you to use external audio effect units in a similar way to using the internal Logic Studio effects. This only makes sense if you are using an audio interface which provides discrete inputs and outputs (analog or digital), used to send signals to and from the external audio effect unit. I/O Parameters  Output Volume field and slider: Adjusts the volume of the output signal.  Output menu: Assigns the respective output (or output pair) of your audio hardware to the plug-in.
Multichannel Gain The Multichannel Gain lets you control the gain (and phase) of each channel of a surround track or bus independently. Â Â Â Â Master slider: Sets the master gain for the combined channel output. Channel gain sliders: Each slider sets the gain for its channel. Phase Invert buttons: When selected, the phase of the selected channel is inverted. Mute buttons: When selected, the channel is muted from the overall output.
 Sine Sweep button: Activate to generate a user-defined frequency spectrum sine wave sweep.  Time field: Determines the duration of the sweep.  Start Freq and End Freq fields: Define the oscillator frequency at the beginning and end of the sine sweep.  Sweep Mode (extended parameter): Choose either a Linear or Logarithmic sweep curve.  Trigger button: Click to trigger the sine sweep.
15 EVOC 20 PolySynth 15 The EVOC 20 PolySynth combines a vocoder with a polyphonic synthesizer, and can be played in real time. The EVOC 20 PolySynth is a sophisticated vocoder, equipped with a polyphonic synthesizer, and capable of receiving MIDI note input. This allows the EVOC 20 PolySynth to be played, resulting in classic vocoder choir sounds, for example. Single notes and chords played with the polyphonic EVOC 20 PolySynth will sing with the articulation of the analysis audio source.
Vocoder Basics If you are new to vocoders you should read this section. It provides you with basic knowledge about vocoders and their functionality. You will also find tips on using vocoders, and achieving good speech intelligibility. What Is a Vocoder? The word vocoder is an abbreviation for VOice enCODER. A vocoder analyses and transfers the sonic character of the audio signal arriving at its analysis input to the audio signal present at its synthesis input.
An envelope follower is coupled to each filter band. The envelope follower of each band tracks (follows) any volume changes in the portion of the audio source allowed to pass by the associated bandpass filter. In this way, the envelope follower of each band generates dynamic control signals. These control signals are then sent to the synthesis filter bank where they control the levels of the corresponding synthesis filter bands. This is done via VCAs—Voltage Controlled Amplifiers.
Using the EVOC 20 PolySynth To make use of the EVOC 20 PolySynth, you need to insert the EVOC 20 PolySynth into an instrument channel’s Instrument slot, and you also need to provide an audio signal as the analysis audio source. You can do this by following these steps: 1 Select or create a new audio track in the Arrange window. 2 Insert (or record) an audio file—use a vocal part to start with—onto this audio track.
EVOC 20 PolySynth Parameters The EVOC 20 PolySynth interface is divided into six main sections. Sidechain Analysis section Formant Filter section Output section Synthesis section Modulation section U/V Detection section  Synthesis section: Controls the polyphonic synthesizer of the EVOC 20 PolySynth. See “Synthesis Parameters” on page 210.  Sidechain Analysis section: The parameters in this section define how the EVOC 20 PolySynth reacts to the analysis signal.
Synthesis Parameters The EVOC 20 PolySynth is equipped with a polyphonic synthesizer. It is capable of accepting MIDI note input. The parameters of the Synthesis section are described below. Mode Buttons These buttons determine the number of voices used by the EVOC 20 PolySynth: Â When Poly is selected, the maximum number of voices is set via the numeric field alongside the Poly button. Note: Increasing the number of voices also increases processor overhead.
Oscillator Section The EVOC 20 PolySynth is equipped with a two oscillator digital synthesizer which features a number of waveforms, and FM (Frequency Modulation). Further to these sound-generators in the Synthesis section is an independent noise generator. Click here to switch between Dual and FM mode There are two oscillator modes. Â Dual: Two oscillators make use of single-cycle digital waveforms to provide the synthesis sound source(s).
Note: When in FM mode, the waveform of Wave 1 is a fixed sine wave. The waveform parameter of Wave 1 does not have an effect in this mode. Wave 2 Parameters The numerical value beside the Wave 2 label indicates the currently selected waveform type. The EVOC 20 PolySynth features 50 single-cycle digital waveforms with different sonic characteristics. To switch between waveforms, do one of the following: m Click-hold on the numerical waveform field and drag up or down.
Dual Mode Parameters The parameters specific to Dual mode are found in the Wave 2 section, and the Balance slider to the right. Â Semi parameter: Adjusts the tuning of the second oscillator (Wave 2) in semitone steps. Â Detune parameter: Fine-tunes Wave 1 and Wave 2 in cents. 100 cents equals a semitone step. Doing so will detune Wave 1 in conjunction with Wave 2 around the tuning zero point. Â Balance slider: Allows you to blend the two oscillator signals (Wave 1 and Wave 2).
Tuning and Pitch Parameters  Analog knob: Simulates the instability of analog circuitry found in vintage vocoders. Analog alters the pitch of each note randomly. This behavior is much like that of polyphonic analog synthesizers. The Analog knob controls the intensity of this random detuning.  Tune: Defines the range of detuning.  Glide: Glide determines the time it takes for the pitch to slide from one note to another (portamento).  Bend Range: Determines the pitch bend modulation range in semitones.
Envelope Parameters The EVOC 20 PolySynth features an Attack/Release envelope generator used for level control of the Oscillator section. Â Attack slider: Determines the amount of time that it takes for the oscillators of the Synthesis section to reach their maximum level. Â Release slider: Determines the amount of time that it takes for the oscillators of the Synthesis section to reach their minimum level.
Release The Release parameter determines how quickly each envelope follower (coupled to each analysis filter band) reacts to falling signals. Longer Release times cause the analysis input signal transients to sustain longer at the vocoder’s output. Note: A long Release time on percussive input signals (a spoken word or hi-hat part, for example) will translate into a less articulate vocoder effect. Note that Release times that are too short result in rough, grainy vocoder sounds.
Formant Filter Parameters The Formant Filter display is divided into two sections by a horizontal line. The upper half applies to the Analysis section and the lower half to the Synthesis section. Changes made to the High and Low Frequency parameters, the Bands parameter, or the Formant Stretch and Shift parameters will result in visual changes to the Formant Filter display. This provides you with invaluable feedback on what is happening to the signal as it is routed through the two formant filter banks.
Lowest and Highest These parameters can be found in the two small fields on either side of the Formant Filter display. These switches determine whether the lowest and highest filter bands act as bandpass filters (like all of the bands between them), or whether they act as lowpass/highpass filters, respectively. Click once on them to switch between the two curve shapes available. Â In the Bandpass setting, the frequencies below/above the lowest/highest bands are ignored for both analysis and synthesis.
Resonance Resonance is responsible for the basic sonic character of the vocoder: low settings give it a soft character, high settings will lead to a more snarling, sharp character. Increasing the Resonance value emphasizes the middle frequency of each frequency band. Note: The use of either, or both, of the Formant Stretch and Formant Shift parameters can result in the generation of unusual resonant frequencies—when high Resonance settings are used.
Intensity and Int via Whl The Intensity slider controls the amount of Formant Shift modulation by the Shift LFO. The Int via Whl slider for the Pitch LFO features a multi-part slider. The intensity of LFO pitch modulation can be controlled by the modulation wheel of an attached MIDI keyboard. The upper half of the slider determines the intensity when the modulation wheel is set to its maximum value, and the lower half when set to its minimum value.
Unvoiced/Voiced (U/V) Detection Human speech consists of a series of voiced sounds (tonal sounds) and unvoiced sounds (noisy sounds). The main distinction between voiced and unvoiced sounds is that voiced sounds are produced by an oscillation of the vocal cords, while unvoiced sounds are produced by blocking and restricting the air flow with lips, tongue, palate, throat, and larynx.
When high settings are used, the increased sensitivity to unvoiced signals can lead to the U/V source—determined by the Mode parameter—being used on the majority of the input signal, including voiced signals. Sonically, this results in a sound that resembles a radio signal which is breaking up, and contains a lot of static or noise. Mode Mode selects the sound sources that can be used to replace the unvoiced content of the input signal. Possible settings are Off, Noise, Noise + Synth, or Blend.
Output Parameters This section covers the various parameters available in the EVOC 20 PolySynth output section. Signal This menu offers the choice of Voc(oder), Syn(thesis), and Ana(lysis). These settings allow you to determine the signal that you wish to send to the EVOC 20 PolySynth main outputs. To hear the vocoder effect, the Signal parameter should be set to Voc. The other two settings are useful for monitoring purposes. Ensemble The three Ensemble buttons switch the ensemble effects on or off.
Block Diagram This block diagram illustrates the signal path in the EVOC 20 TrackOscillator and EVOC 20 PolySynth. Analysis source Analysis section Legend Track -----------Side chain Audio signal Control signal R L Stereo to mono Parameter control Sensitivity U/V detection Frequency range between highest/lowest 1-5 Filter bank with five bands (example) TO: pitch analysis Envelope follower 1-5 A Freeze B Synthesis section TO: Max/Quant.
Editing the Analysis and Synthesis Signals The following section outlines how you can edit the analysis and synthesis signals to achieve better speech intelligibility. Compressing the Analysis Signal The less the level changes, the better the intelligibility of the vocoder. You should therefore compress the analysis signal in most cases. Enhancing High Frequency Energy The vocoder, in a way, always generates the intersection point of the analysis and synthesis signals.
Gating Background Noises in the Analysis Signal If the analysis signal is compressed, as recommended, the level of breath, rumble, and background noises will rise. These background noises can cause the vocoder bands to open, but this is normally not intended. In order to eliminate these noises, it’s therefore a good idea to employ a noise gate before compression and treble boosting. If the analysis signal is gated appropriately, you may find that you want to reduce the Analysis Release value.
Feel free to do what you like when setting the Formant parameters. The intelligibility of speech is surprisingly little affected by shifting, stretching, or compressing the formants. Even the number of frequency bands used has a minimal impact on the quality of intelligibility. The reason for this is our ability to intuitively differentiate the voices of children, women, and men, whose skulls and throats vary vastly by nature.
Werner Meyer-Eppler, the director of Phonetics at Bonn University, recognized the relevance of the machines to electronic music after Dudley visited the University in 1948. Meyer-Eppler used the vocoder as a basis for his future writings which, in turn, became the inspiration for the German “Elektronische Musik” movement. In the 1950s, a handful of recordings ensued. In 1960, the Siemens Synthesizer was developed in Munich. Among its many oscillators and filters, it included a valve-based vocoding circuit.
16 EFM1 16 The 16-voice EFM1 is a simple, but powerful, frequency modulation synthesizer. It produces the typically rich bell and digital sounds that frequency modulation (FM) synthesis has become synonymous with. At the core of the EFM1 engine, you’ll find a multi-wave modulator oscillator and a sine wave carrier oscillator. The Modulator oscillator modulates the frequency of the carrier oscillator within the audio range, thus producing new harmonics. These harmonics are known as sidebands.
 The bottom section houses the Output section, and features the Sub Osc Level and Stereo Detune parameters, plus the volume envelope, Main Level, and Velocity controls. A Randomize field is shown to the lower right.  The extended parameters panel (accessed by clicking the disclosure triangle at the lower left) allows you to assign MIDI controllers to the FM Amount (FM depth, in other words) and Vibrato parameters.
Modulator and Carrier The modulator and carrier parameters are outlined below. Harmonic In FM synthesis, the basic overtone structure is determined by the tuning relationship of the modulator and volume envelope. This is often expressed as a tuning ratio. In the EFM1, this ratio is achieved with the Modulator and Carrier Harmonic controls. Additional tuning control is provided by the Fine (Tune) parameters. You can tune the modulator and volume envelope to any of the first 32 harmonics.
FM Parameters These parameters affect the frequency modulation aspects of the EFM 1. FM (Intensity) The modulator oscillator modulates the volume envelope frequency, resulting in newly generated sidebands that add new overtones. Turning up the FM (Intensity) control (the large dial in the center) produces increasing numbers of overtones—and the sound becomes brighter. The FM (Intensity) parameter is sometimes called the FM Index.
LFO The LFO (Low Frequency Oscillator) serves as a cyclic modulation source for FM Intensity or Vibrato. Turning the LFO control clockwise increases the effect of the LFO on FM Intensity. Turning it counter clockwise introduces a vibrato. In the center (0) position the LFO does not have an effect. You can easily center the LFO dial by clicking on the 0. Rate The speed/rate of the LFO cycles is set with the Rate parameter. The Output Section The EFM1 provides several level controls, as discussed below.
Velocity The EFM1 is able to respond to MIDI velocity, and reacts with dynamic sound and volume changes—harder playing will result in a brighter and louder sound. The sensitivity of the EFM1 in response to incoming velocity information is determined by the Velocity parameter. Set the Velocity control all the way to the left (counter-clockwise) if you don’t want the EFM1 to respond to velocity.
17 ES E 17 This chapter discusses the eight-voice polyphonic ES E synthesizer. The ES E (ES Ensemble) is designed for pad and ensemble sounds. It is great for adding atmospheric beds to your music, with minimal CPU overhead. All ES E parameters are discussed in the following section. Â 4, 8, 16 buttons: Determine the ES E’s octave transposition.
 Resonance knob: Sets the resonance of the ES E’s dynamic lowpass filter.  AR Int knob: The ES E features one simple envelope generator per voice—offering an Attack and a Release parameter. The AR Int parameter defines the amount of cutoff frequency modulation (applied by the envelope generator).  Velo Filter knob: Sets the velocity sensitivity of the cutoff frequency modulation (applied by the envelope generator). This parameter has no effect if AR Int is set to 0.
18 ES M 18 The monophonic ES M (ES Mono) is a good starting point if you’re looking for bass sounds that punch through your mix. The ES M compact synthesizer features an automatic fingered portamento mode, making bass slides easy. It also features an automatic filter compensation circuit that delivers rich, creamy basses, even when using higher resonance values. All ES M parameters are discussed in the following section. Â 8, 16, 32 buttons: Set the ES M’s octave transposition.
 Decay (Filter) knob: Sets the decay time of the filter envelope. It is only effective if Int is not set to 0.  Velo (Filter) knob: Determines the velocity sensitivity of the filter envelope. This parameter is only effective if Int is not set to 0.  Decay (Volume) knob: Sets the decay time of the dynamic stage. The attack, release, and sustain times of the synthesizer are internally set to 0.  Velo (Volume) knob: Determines the velocity sensitivity of the dynamic stage.
19 ES P 19 This chapter introduces you to the eight-voice polyphonic ES P (ES Poly) synthesizer. Functionally, (despite its velocity sensitivity) this flexible synthesizer is somewhat reminiscent of the affordable polyphonic synthesizers produced by the leading Japanese manufacturers in the 1980s: Its design is easy to understand, it is capable of producing lots of useful musical sounds, and you may be hard-pressed to make sounds with it that can’t be used in at least some musical style.
 Speed knob: Controls the rate of the oscillator frequency or cutoff frequency modulation.  Frequency knob: Sets the cutoff frequency of the resonance-capable dynamic lowpass filter.  Resonance knob: Sets the resonance of the dynamic lowpass filter. Increasing the Resonance value results in a rejection of bass (low frequency energy) when using lowpass filters. The ES P compensates for this side-effect internally, resulting in a more bassy sound.
20 ES1 20 This chapter introduces the virtual analog ES1 synthesizer. The ES1’s flexible tone generation system and interesting modulation options place an entire palette of analog sounds at your disposal: punchy basses, atmospheric pads, biting leads, and sharp percussion.
2', 4', 8', 16', 32' Buttons These footage values allow you to switch the pitch in octaves. 32 feet is the lowest, and 2 feet, the highest setting. The origin of the term feet to measure octaves, comes from the measurements of organ pipe lengths. Wave Wave allows you to select the waveform of the oscillator, which is responsible for the basic tone color. You can freely set any pulse width in-between the square wave and pulse wave symbols.
Filter Parameters This section outlines the filter parameters available to the ES1. Drive This is an input level control for the lowpass filter, which allows you to overdrive the filter. Its use changes the behavior of the Resonance parameter, and the waveform may sound distorted. Cutoff and Resonance The Cutoff parameter controls the cutoff frequency of the ES1’s lowpass filter. Resonance emphasizes the portions of the signal which surround the frequency defined by the Cutoff parameter.
ADSR via Vel The main envelope generator (ADSR) modulates the cutoff frequency over the duration of a note. The intensity of this modulation can be set to positive or negative values, and can respond to velocity information. If you play pianissimo (Velocity = 1), the modulation will take place as indicated by the lower arrow. If you strike with the hardest fortissimo (Velocity = 127), the modulation will take place as indicated by the upper arrow.
All ADSR parameters will always remain active for the filter (ADSR via Vel). A stands for attack time, R for release time, while Gate is the name of a control signal used in analog synthesizers, which tells an envelope generator that a key is pressed. As long as an analog synth key is pressed, the gate signal maintains a constant voltage. Used as a modulation source in the voltage controlled amplifier (instead of the envelope itself ), it creates an organ type envelope without any attack, decay, or release.
    The mix between the main and sub oscillators The cutoff frequency of the filter The resonance of the filter The main volume (the amplifier) The following two targets are only available for the modulation envelope:  Filter FM (the amount of cutoff frequency modulation by the triangle wave of the oscillator): The modulation characteristics are nonlinear.
Tune Tune sets the pitch of the ES1. Analog Analog slightly alters the pitch of each note, and the cutoff frequency, in a random manner. Similar to polyphonic analog synthesizers, Analog values higher than zero allow the oscillators of all triggered voices to cycle freely. Note that if Analog is set to a value of zero, the oscillator cycle start points of all triggered voices are synchronized. This may be useful for percussive sounds, when looking to achieve a sharper attack characteristic.
MIDI Controller List 248 Controller Number Parameter Name 12 Oscillator pitch buttons 13 Oscillator waveform 14 Mix slider 15 Waveform of sub oscillator 16 Drive slider 17 Cutoff slider 18 Resonance slider 19 Slope buttons 20 ADSR via Vel: lower slider 21 ADSR via Vel: upper slider 22 Attack slider 23 Decay slider 24 Sustain slider 25 Release slider 26 Key slider 27 Amplifier Envelope Selector buttons 28 Level via Velocity: lower slider 29 Level via Velocity: upper sli
21 ES2 21 The ES2 synthesizer combines a powerful tone generation system with extensive modulation facilities. The ES2 provides three oscillators, which can be synchronized and ring-modulated. Pulse-width modulation is also possible. Oscillator 1 can be modulated in frequency by Oscillator 2, and is capable of producing FM-style synthesizer sounds. In addition to the classic analog synthesizer waveforms, the ES2 oscillators also provide 100 single-cycle waveforms, known as Digiwaves.
The ES2 Parameters If given just a few words to explain the principles behind a subtractive synthesizer, it would go something like this: The oscillator generates the oscillation (or waveform), the filter takes away the unwanted overtones (of the waveform), and the dynamic stage sets the volume of the permanent oscillation (the filtered waveform) to zero—as long as no keyboard note is pressed.
Global Parameters These parameters impact on the overall instrument sound produced by the ES2. You can find the global parameters to the left of the oscillators, and above the filter section. In surround instances, two additional global parameters are shown in the Extended Parameters section. Global parameters Global parameters Tune Tune sets the pitch of the ES2 in cents. 100 cents equals a semitone step. At a value of 0 c (zero cents), a' is tuned to 440 Hz or concert pitch.
The CBD (Constant Beat Detuning) parameter matches this natural effect by detuning the lower frequencies in a ratio proportionate to the upper frequencies. Besides disabling CBD altogether, four values are at your disposal: 25%, 50%, 75%, 100%. If you choose 100%, the phasing beats are (almost) constant across the entire keyboard range. This value, however, may be too high, as the lower notes might be overlydetuned at the point where the phasing of the higher notes feels right.
Note: If you switch to Legato, you need to play legato to actually hear the Glide parameter taking effect. Note: On several monophonic synthesizers, the behavior in Legato mode is referred to as single trigger, while Mono mode is referred to as multi trigger. Voices The maximum number of notes that can be played simultaneously is determined by the Voices parameter. Maximum value for Voices is 32.
 If Osc Start is set to hard, each initial oscillator phase begins at the highest possible level in its waveform cycle every time a note is played. This punch is only audible if the ENV3 Attack Time is set to a minimal value—a very fast attack. This setting is highly recommended for electronic percussion and hard basses. Note: Osc Start soft and hard result in a constant output level of the initial oscillator phase every time the sound is played back.
Muting Oscillators By clicking on the green numbers to the right of the oscillators, you can mute and unmute them independently. This saves processor power. Frequency Knobs The Frequency knobs set the pitch in semitone steps over a range of ±3 octaves. As an octave consists of 12 semitones, the ±12, 24, and 36 settings represent octaves. You can click on these options to quickly set the corresponding octave.
The sine wave can be modulated in frequency by Oscillator 2 in the audio frequency range. This kind of linear frequency modulation is the basis on which FM synthesis works. FM synthesis was popularized by synthesizers such as Yamaha’s DX7 (the architecture of which is much more complex, when it comes to FM synthesis). A click on the oscillator number disables the output of Oscillator 1.
Linear Frequency Modulation The principle of linear frequency modulation (FM) synthesis was developed in the late sixties and early seventies by John Chowning. It’s such a flexible and powerful method of tone generation that synthesizers were developed, based solely on the idea of FM synthesis. The most popular FM synthesizer ever built is Yamaha’s DX7. FM synthesis is also found in other models of the Yamaha DX range and some Yamaha E-Pianos.
The Waveforms of Oscillators 2 and 3 Basically, Oscillators 2 and 3 supply the same selection of analog waveforms as Oscillator 1: sine, triangular, sawtooth, and rectangular waves. The pulse width can be scaled steplessly between 50% and the thinnest of pulses, and can be modulated in a number of ways (see the “Pulse Width Modulation” on page 258).
Sync The rectangular and sawtooth waveforms also feature a Sync option. In this mode, the frequency of Oscillator 2 (or 3, respectively) is synchronized to the frequency of Oscillator 1. This does not mean that their frequency controls are simply disabled. They still oscillate at their selected frequencies, but every time that Oscillator 1 starts a new oscillation phase, the synchronized oscillator is also forced to restart its phase from the beginning.
White and Colored Noise (Oscillator 3 Only) Unlike Oscillator 2, Oscillator 3 is not capable of producing ring modulated signals or sine waves. Its sonic palette however, is bolstered by the inclusion of a noise generator. By default, Oscillator 3’s noise generator generates white noise. White noise is defined as a signal that consists of all frequencies (an infinite number) sounding simultaneously, at the same intensity, in a given frequency band. The width of the frequency band is measured in Hertz.
Note that the vector envelope features a loop function. This addition extends its usefulness, allowing you to view it as a luxurious pseudo-LFO with a programmable waveform. It can be used for altering the positioning of the Triangle and Square cursors. Read more about this in “Vector Mode Menu” on page 292, and “The Vector Envelope” on page 293.
Filters The ES2 features two dynamic filters which are equivalent to the Voltage Controlled Filters (VCF) found in the world of analog synthesizers. The two filters are not identical. Filter 1 features several modes: lowpass, highpass, bandpass, band rejection, peak. Filter 2 always functions as a lowpass filter. Unlike Filter 1, however, Filter 2 offers variable slopes (measured in dB/octave). Filter button The Filter button bypasses (switches off ) the entire filter section of the ES2.
The mono output signal of Filter 2 is then fed into the input of the dynamic stage (the equivalent of a VCA in an analog synthesizer), where it can be panned in the stereo or surround spectrum, and then fed into the effects processor. In the graphic to the right, the filters are cabled in parallel. If Filter Blend is set to 0, you’ll hear a 50/50 mix of the source signal routed via Filter 1 and Filter 2, which is fed into the mono input of the dynamic stage.
In conjunction with the overdrive/distortion circuit (Drive) and a series cabling configuration, the ES2’s signal flow is far from commonplace. The graphics illustrate the signal flow between the Oscillator Mix stage (the Triangle) and the dynamic stage, which is always controlled by ENV 3. The signal flow through the filters, the overdrives and the sidechains is dependent on the Filter Blend setting. Filter 1 Mix Drive Filter 2 Filter Blend in parallel filter mode.
Drive The filters are equipped with separate overdrive modules. Overdrive intensity is defined by the Drive parameter. If the filters are connected in parallel, the overdrive is placed before the filters. If the filters are connected in series, the position of the overdrive circuits depends on the Filter Blend parameter—as described above. Polyphonic Distortions in the Real World The ES2 features a distortion effect, equipped with a tone control, in the Effects section.
     Filter 1 set to Peak Filter mode High Resonance value for Filter 1 Modulate Cutoff Frequency 1 manually or in the Router. Set Drive to your taste. Filter away (cut) the high frequencies with Filter 2 to taste. The sonic result resembles the effect of synchronized oscillators. At high resonance values, the sound tends to scream. Modulate the Resonance of Filter 1, if you wish. Filter Parameters This section covers the ES2 filter parameters in detail.
The Chain Symbols Manipulating the Cutoff and Resonance controls in real time is one of the key ingredients in the creation of expressive synthesizer sounds. You’ll be pleased to know that you can control two filter parameters at once by dragging on one of the three little chain symbols in the filter graphic. Â The chain between Cut and Res of Filter1 controls Resonance (horizontal mouse movements) and Cutoff (vertical mouse movements) of the first filter simultaneously.
Filter Slope A filter can not completely suppress the signal portion outside the frequency range defined by the Cutoff Frequency parameter. The slope of the filter curve expresses the amount of rejection applied by the filter (beneath the cutoff frequency) in dB per octave. Filter 2 offers three different slopes: 12 dB, 18 dB and 24 dB per octave. Put another way, the steeper the curve, the more severely the level of signals below the cutoff frequency are affected in each octave.
 The abbreviation BP stands for bandpass. In this mode, only the frequency band directly surrounding the cutoff frequency can pass. All other frequencies are cut. The Resonance parameter controls the width of the frequency band that can pass. The bandpass filter is a two-pole filter with a slope of 6 dB/octave on each side of the band. Filter 2 FM The cutoff frequency of Filter 2 can be modulated by the sine wave of Oscillator 1, which means that it can be modulated in the audio frequency range.
Handling Processing Power Economically The ES2 is designed to make the most efficient use of computer processing power. Modules and functions that are not in use don’t use processing power. This principle is maintained by all elements of the ES2. As examples: If only one of the three oscillators is in use, and the others are muted, less processing power is required. If you do not modulate Digiwaves, or if you disengage the filters, processing power is saved.
Sine Level The Sine Level knob (located next to the Filter 2 section) allows the mixing of a sine wave (at the frequency of Oscillator 1) directly into the dynamic stage, independent of the filters. Even if you have filtered away the basic partial tone of Oscillator 1 with a highpass filter, you can reconstitute it through use of this parameter.
Ten such modulations of source, via, and target can take place simultaneously, in addition to those which are hard-wired outside the Router. The bypass (b/p) parameter allows the disabling/enabling of individual modulation routings without losing settings. Note: Some modulations aren’t possible, due to technical reasons. As an example, the envelope times can be modulated by parameters that are only available during a noteon message.
A Modulation Example Say you’ve chosen these settings:  Target: Pitch 123  via: Wheel  Source: LFO1  Modulation intensity: Slider position, set as desired In this configuration, the modulation source—LFO1—is used to modulate the frequency (pitch) of all three Oscillators (Pitch 123). (Pitch 123) is the modulation target in this example. You’ll hear a vibrato (a modulation of the pitch) at the speed of LFO 1’s Rate.
Detune This target controls the amount of detuning between all three oscillators. Note: The sensitivity of all the pitch modulation targets described above depends on the modulation intensity. This sensitivity scaling allows for very delicate vibrati in the cent range (one cent equals 1/100 semitone), as well as for huge pitch jumps in octave ranges. Â Â Â Â Â Â Modulation intensity from 0 to 8: steps are 1.25 cents. Modulation intensity from 8 to 20: steps are 3.33 cents.
Osc1Wave Dependent on the waveform selected, you can control the pulse width of rectangular and pulse waves of Oscillator 1, the amount of frequency modulation (with Oscillator 1 being the carrier and Oscillator 2 being the modulator), or the position of the Digiwave. The pulse width of the rectangular and pulse waves is not restricted to two fixed values in Oscillator 1. Note: In classic FM synthesizers, the amount of FM is controlled in real time by velocity sensitive envelope generators.
SineLevl SineLevl (Sine Level) allows modulation of the sine wave level of Oscillator 1, which can be mixed directly into the input of the dynamic stage—without being affected by the filters. The parameter defines the level of the first partial tone of Oscillator 1. See “Sine Level” on page 271. OscLScle OscLScle (Osc Level Scale) allows modulation of the levels of all three oscillators simultaneously.
Cut1inv2 Cut1inv2 (Cutoff 1 normal and Cutoff 2 inverse) simultaneously modulates the Cutoff frequencies of the first and second filters inversely (in opposite directions). Put another way, while the first filter’s Cutoff frequency is rising, the Cutoff of the second filter will fall—and vice versa. Note: In cases where you have combined Filter 1, defined as a highpass filter, and Filter 2 (which always works in lowpass mode) in Serial mode, both will serve as a bandpass filter.
Lfo1Asym Lfo1Asym (Lfo1 Asymmetry) can modulate the selected waveform of LFO 1. In the case of a square wave, it changes its pulse width. In the case of a triangle wave, it sweeps between triangle and sawtooth. In the case of a sawtooth wave, it shifts its zero crossing. Lfo1Curve This target modulates the waveform smoothing of the square and random wave. In the case of a triangle or sawtooth wave, it changes their curves between convex, linear, and concave.
Env3Atck Env3Atck (Envelope 3Atck) modulates the Attack time of the third envelope generator. Env3Dec Env3Dec (Envelope 3 Decay) modulates the Decay time of the third envelope generator. Env3Rel Env3Rel (Envelope 3 Release) modulates the Release time of the third envelope generator. Env3Time Env3Time (Envelope 3 All Times) modulates all of ENV3’s time parameters: Attack time, Decay time, Sustain time, and Release time. Glide This target modulates the duration of the Glide (portamento) effect.
Pad-X, Pad-Y These modulation sources allow you to define the axes of the Square, for use with the selected modulation target. The cursor can be moved to any position in the Square, either manually or controlled by the vector envelope. See “The Square” on page 291 and “The Vector Envelope” on page 293. Max If you select Max as a source, the value of this source will permanently be set to +1.
Whl+To The modulation wheel and aftertouch serve as modulation sources. MIDI Controllers A–F MIDI controllers available in the mod matrix are named Ctrl A–F and can be assigned to arbitrary controller numbers (via the MIDI Controllers Assignment menus at the bottom of the ES2 interface). Note: Earlier ES2 versions offered the Expression, Breath, and MIDI Control Change Messages 16 to 19 as modulation sources.
RndN02 RndNO2 (Note On Random2) behaves like Note On Random1 with the exception that it glides to the new random value using the Glide time (inclusive of modulation). It also differs from NoteOnRandom1 in that the (random modulation) value changes when playing legato, while in legato mode. SideCh SideCh (Side Chain modulation) uses a Side Chain (tracks, inputs, busses) to create a modulation signal. The Side Chain source can be selected in the upper gray area of the window.
If you select Pitch 123 as the target, modulate it with the LFO1 source, and select Keyboard as the via value, the vibrato depth will change, dependent on key position. Put another way, the vibrato depth will be different for notes higher or lower than the defined Keyboard position. Velo If you select Velo (Velocity) as the via value, the modulation intensity will be velocity sensitive—modulation will be more or less intense dependent on how quickly (hard) you strike the key.
All parameters that allow you to select a MIDI controller feature a Learn option. If this is selected, the parameter will automatically be assigned by the first appropriate incoming MIDI data message. The Learn mode features a 20 second time-out facility: If the ES2 does not receive a MIDI message within 20 seconds, the parameter will revert to its original MIDI controller assignment. Note: As the new entry is added to the top of the list, existing automation data is increased by one.
The LFOs LFO is the abbreviated form of Low Frequency Oscillator. In an analog synthesizer, LFOs deliver modulation signals below the audio frequency range—in the bandwidth that falls between 0.1 and 20 Hz, and sometimes as high as 50 Hz. LFOs serve as modulation sources for periodic, cyclic modulation effects. If you slightly modulate the pitch of an audio oscillator at a rate (speed, LFO frequency) of, say, 3–8 Hz, you’ll hear a vibrato.
Most commonly, this is used for delayed vibrato—many instrumentalists and singers intonate longer notes this way. To set up a delayed vibrato: Place the slider at a position in the upper half (Delay) and modulate the Pitch123 target with the LFO1 source. Set a slight modulation intensity. Select a Rate of about 5 Hz and the triangular wave as the LFO waveform.
The term Sample & Hold (abbreviation—S & H) refers to the procedure of taking samples from a noise signal at regular intervals. The voltage values of these samples are then held until the next sample is taken. When converting analog audio signals into digital signals, a similar procedure takes place: Samples of the voltage of the analog audio signal are taken at the rate of the sampling frequency.
Unlike many other synthesizers, there is no hard-wired connection between any of the envelope generators and the cutoff frequencies of the ES2 filters. Modulation of the cutoff frequencies must be set separately in the Router. In the default setting, this is already the case—in the Router channel just below the Filter (see graphic). Set up a Router channel as follows, in order to establish this type of modulation: Set target to Cutoff 1, Cutoff 2, or Cut 1+2, set source to, say, ENV 2.
 In Retrig mode, the envelope will be triggered by any key you strike, no matter whether other notes are sustained or not. Every sustained note is affected by the retriggered envelope. The design of early analog polysynths led to polyphonic instruments where all voices passed through a single lowpass filter. This design was primarily due to cost factors. The best known example of these polyphonic instruments were the Moog Polymoog, the Yamaha SK20, and Korg Poly 800.
The Parameters of ENV 2 and ENV 3 The feature sets of ENV 2 and ENV 3 are identical, but it is always the task of ENV 3 to define the level of each note—to modulate the dynamic stage. ENV 3 is available for simultaneous use as a source in the Router as well. The envelope’s time parameters can also be used as modulation targets in the Router. Note: See “Envelopes” on page 626 for information on the basic functionality and meaning of envelope generators.
In this position, the Sustain level (abbreviated as S) defines the level that is sustained for as long as the key remains depressed, following the completion of the Attack time and Decay time phases. The Sustain Time slider defines the time it takes for the level to rise to its maximum—or to fall to zero—after the decay phase is over. Settings in the lower half of its range (Fall) determine the speed at which the level decays from the Sustain level to zero.
As an alternative to this real-time control, the position of the cursor can be modulated by the vector envelope—just like the mix between the three oscillators in the Triangle. The Loop function of the vector envelope generator allows for cyclic movements. This opens a number of doors, allowing it to operate as a two-dimensional, luxurious pseudo-LFO with a programmable waveform. More on this is found in “The Vector Envelope” on page 293.
The Vector Envelope The Triangle and Square are the most special and unusual elements of the ES2’s graphical user interface. Whilst the Triangle controls the mix of the three oscillators, the X and Y axes of the Square can modulate any (modulation) target. Triangle Square Vector envelope The vector envelope can control the movement of the cursors in the Triangle and the Square in realtime.
Note: A number of vector envelope editing commands can be quickly accessed in a shortcut menu. Control-click anywhere on the vector envelope to open it. Sustain Point Any point can be declared the sustain point. Given that the note played is sustained long enough and there’s no loop engaged, any envelope movement will stop when this sustain point is reached. It will be sustained until the key is released (until the MIDI note-off command).
In order to define a point as the loop point, click on the turquoise strip below the desired point. A loop point is indicated by an L in the strip below. In order to see and define the loop point, the loop must be activated. See “Loop Mode” on page 298. ∏ Tip: With loop activated, the vector envelope works like a multi-dimensional, polyphonic LFO with a programmable waveform.
 You can disable the vector envelope altogether (or only the Triangle or Square) as described in the “Vector Mode Menu” on page 292. Setting and Deleting Points The more points you set, the more complex the vector envelope movements that can be designed. You can:  Create a new point by Shift-clicking between two existing points. The segment that previously existed between the two old points is divided at the mouse position.
Env Modes Normal and Finish If the Env Mode menu is set to Normal, the release phase (the phase after the sustain point) will begin as soon as you release the key (note off ). The release phase will start from the vector envelope point where you released the key. Â If the loop is switched off, and the vector envelope reaches the sustain point, the sustain point will be played for as long as you hold the key.
Loop Mode The ES2 features these loop modes: Â Off: If the Loop Mode mode is set to Off, the vector envelope runs in one shot mode from its beginning to its end—given that the note is held long enough. The other loop parameters are disabled. Â Forward: When Loop Mode is set to Forward, the vector envelope runs to the sustain point and begins to repeat the section between the loop point and sustain point periodically, always in a forward direction.
Loop Smooth When Loop Mode is set to Forward or Backward, there will inevitably be a moment when a transition from the sustain point to the loop point occurs. In order to avoid abrupt cursor position changes, this transition can be smoothed through use of the Loop Smooth parameter. Â If Loop Rate is set to Sync or Free, the loop smoothing time will be displayed as a percentage of the loop cycle duration. Â If Loop Rate is set to as set, the loop smoothing time will be displayed in milliseconds (ms).
Effect Processor The ES2 is equipped with an integrated effect processor. Any changes to this processor’s effects settings are saved as an integral part of each sound program. Despite the inclusion of this integrated effects processor, please feel free to process the ES2 with the other effect plug-ins included in Logic Studio. The sound and parameter set of the integrated effects unit is reminiscent of classic pedal effects, designed for the electric guitar.
Using Controls and Assigning Controllers The section at the bottom of the ES2 interface provides three modes, accessed by clicking the buttons on the left: Â Macro: Shows a number of macro parameters which affect groups of other parameters. Â MIDI: Allows you to assign MIDI controllers to particular Router channels (see “MIDI Controllers A–F” on page 281). Â Macro Only: Replaces the ES2 interface with a dedicated (and much smaller) view that is limited to the macro parameters.
Random Sound Variations The ES2 offers a unique feature that allows you to vary the sound parameters randomly. You can define the amount of random variation, and can restrict the variations to specific sonic elements. The random sound variation feature will inspire and aid (or occasionally amuse) you when creating new sounds. Clicking the RND button randomly alters the sound. The process is triggered by a single click and can be repeated as often as you like.
You can restrict the random sound variation to the parameter groups listed below: All All ES2 parameters, with the exception of the parameters mentioned above, will be altered. All except Router and Pitch All ES2 parameters, with the exception of all Router parameters and the basic pitch (semitone settings of the oscillators), will be altered. The oscillator fine tuning will be varied. This will result in more musically useful sounds.
Vector Env Mix Pad The Oscillator mix levels (Triangle cursor positions) of the vector envelope points are altered. The rhythm and tempo of the modulation (the time parameters of the points) will not be altered. Vector Env XY Pad Options The Square cursor positions of the vector envelope points are altered. The XY routing won’t be altered. The rhythm and tempo of the modulation (the time parameters of the points) will not be altered.
Tutorials You will find the settings for these tutorials in the Tutorial Settings folder in the Settings menu (in the header of the ES2 window). Sound Workshop The Sound Workshop will guide you—from scratch—through the creation of commonly used sounds. The following tutorial section will also guide you through the sound creation process, but starts you off with a number of templates.
 Engage Unison mode and select a higher setting for Analog. As the sound is polyphonic, each note is doubled. The number of notes that can be played simultaneously will be reduced from 10 to 5. This will make the sound rich and broad. Combining Unison and higher values for Analog will spread the sound across the stereo or surround spectrum. In many factory settings, the Unison mode is active. This demands a lot of processing power.
Clean Bass Settings With One Oscillator Only Not every sound needs to comprise of several oscillators. There are numerous simple, effective, sounds which make use of a single oscillator. This is especially true of synthesizer bass sounds, which can be created quickly and easily with the basic “Analog Bass clean” setting. The basic sound is a rectangular wave, transposed down by one octave. The sound is filtered by Filter 2. What’s special about this sound is its combination of Legato and Glide (portamento).
 Alter the modulator frequency (Oscillator 2) by adjusting Fine Tune from 0 c to 50 c. You’ll hear a very slow frequency modulation, that can be compared to the effect of an LFO. The frequency modulation, however, takes place in the audio spectrum. It is adjusted in semitone steps by the frequency selector. Check out the entire range from –36 s to +36 s for Oscillator 2. You’ll hear a broad spectrum of FM sounds. Some settings will remind you of classic FM synthesizer sounds.
FM With Digiwaves In the FM Digiwave setting, a Digiwave is used as an FM modulator. This results in a bell-like spectra from only two operators. Using traditional FM synthesis, this type of timbre could normally only be produced with a larger number of sine oscillators. In order to create a fatter, undulating, and atmospheric quality to the sound, the polyphonic Unison mode has been engaged. Filter and amplitude envelopes have been preset to shape the sound.
You can further develop the sound by applying filtering, envelope modulations, and effects. There is, however, one small problem—the sound is out of tune. Â Use Oscillator 3 as a reference for the tuning of the FM sound, by moving the cursor in the Triangle. Â You’ll notice that the sound is 5 semitones too high (or 7 semitones too low, respectively). Â Transpose both oscillators 1 and 2 five semitones (500 ct) lower.
 Reduce the Cutoff Frequency and Resonance of Filter 1 to make the sound softer.  Save the new setting.  Compare the result with the PWM 2 Osc setting. You’ll hear that the sound has undergone a remarkable evolution.  Compare it to PWM Soft Strings, which was created as described above. You’ll probably notice a few similarities. Ring Modulation A ring modulator takes its two input signals and outputs the sum and difference frequencies of them.
 In the second Router channel, an envelope pitch modulation has been preprogrammed (target = Pitch 2, Source = Env 1). Setting the minimum value to 1.0 results in a typical sync envelope. Also check out shorter Decay Times for Envelope 1.  In order to avoid a sterile, lifeless sound (after the decay phase of the envelope), you may also want to modulate the oscillator frequency with an LFO. Use the third Router channel: set the minimum modulation applied by LFO 1 to about 0.50.
Vector Synthesis—XY Pad The vector envelope example starts where the first one left off. You have a simple vector envelope consisting of 4 points, which is set to modulate the oscillator mix (the Triangle). In this example, the vector envelope will be used to control two additional parameters: The Cutoff Frequency of Filter 2 and Panorama. These are pre-set as the X and Y targets in the Square. Both have a value of 0.50.
The distances between the points of the vector envelope have been set to be rhythmically exact. Given that Loop Rate has been engaged, the time values are not displayed in ms, but as percentages. There are four time values (each at 25%), which is a good basis for the transformation into note values. Â Switch off the vector envelope by setting Solo Point to on. This allows you to audition the individual points in isolation.
Percussive Synthesizers and Basses With Two Filter Decay Phases As with the Vector Kick, the Vector Perc Synth setting uses the vector envelope to control the filter cutoff frequency (with two independently adjustable decay phases). This would not be possible with a conventional ADSR envelope generator.
Have a look at its architecture: Osc 1 and 3 provide the basic wave combination within the Digiwave field. Changing the Digiwaves of both (in combination) delivers a huge number of basic variations— some also work pretty well for electric piano-type keyboard sounds. Osc 2 adds harmonics with its synced waveform, so you should only vary its pitch or sync waveform. There are a couple of values which can be changed here, which will give you a much stronger, more balanced signal.
Velocity is set up to be very responsive because many synthesizer players don’t strike keys like a piano player would with a weighted-action “punch.” As such, you should play this patch softly, or you may find that the slap tends to sweep a little. Alternately, you can adjust the sensitivity of the filter modulation’s velocity value to match your personal touch.
Please feel free to find your own values. While doing so, keep in mind the fact that there are two modulation couples, which should only be changed symmetrically (Mod. 2 and 3 work as a pair of twins, and also Mod. 6 and 7). So, if you change Pitch 2’s maximum to a lower minus value, remember to set Pitch 3’s maximum value to the same positive amount (same goes for modulation pair 6 and 7). You can also bring in LFO 2 to increase the pitch diffusion against LFO 1’s pitch and pan movements.
 A keyboard assignment was set up as the Modulation 4 source. This is because all pitch, or pulse-width, modulations tend to cause a stronger detuning in the lower ranges, while the middle and upper key zones feature the desired diffusion effect. When using this parameter, you should initially adjust the lower ranges until an acceptable amount of detuning (resulting from the modulation) is reached. Once set, check whether or not the modulations in the upper zones work to your satisfaction.
MW-Pad-Creator 3 This is an attempt to create a patch which is able to automatically generate new patches. The Basics Again, Oscillator 2 is used for a pulse width modulation—which creates a strong ensemble component (please refer to “Something Horny (Crescendo Brass)” on page 318, for further information). Oscillators 1 and 3 are set to an initial start wave combination within their respective Digiwave tables.
Another Approach to Crybaby (Wheelsyncer) Never obsolete—and undergoing a renaissance in new popular electronic music: Sync Sounds The technical aspects of forcing an Oscillator to sync are described in “Sync” on page 259. Here’s the practical side of the playground. Wheelsyncer is a single-oscillator lead sound, all others are switched off. Although Oscillator 2 is the only one actively making any sound, it is directly dependent on Oscillator 1.
22 EVB3 22 The EVB3 software instrument mimics the sound and features of the Hammond B3 and Leslie sound cabinet. The EVB3 simulates an organ with two manuals (keyboards) and a pedalboard—each of which can have its own registration (sound settings). The sound generation process fully simulates the tone wheel generators of an electro-mechanical Hammond organ, down to the smallest detail.
MIDI Setup If you want to fully exploit all features of the EVB3, you will need a MIDI (bass) Pedal unit, and two 73-key MIDI keyboards. As the EVB3 also emulates the B3’s preset keys, the lowest octave of attached MIDI keyboards can switch the EVB3 registrations, just like the original B3. Please read the following section for more information. The EVB3 can, of course, be played with single-manual keyboards with the standard 61 keys (5 octaves C to C).
Changing MIDI Channels You also can set the EVB3 to receive on MIDI channels other than its default configuration. This is done with the Basic MIDI Ch parameter in the Controls view’s General section. This parameter assigns a MIDI channel to the Upper manual. The receive channel number for Lower is always one channel number higher than the channel assigned to the Upper manual. The Pedal register receive channel is always two channel numbers higher than the channel assigned to the Upper manual.
Keyboard Split The EVB3 can also be played perfectly with a single MIDI keyboard (one manual) that is only capable of transmitting on one MIDI send channel. You can split the keyboard in order to play Upper, Lower, and Pedal sounds on different keyboard zones. In the parameter field in the bottom center of the interface, set Keyboard Mode to Split. Set the keyboard zones with the UL Split and LP Split parameters, in conjunction with the Set buttons. The abbreviations are for: Upper/Lower and Lower/Pedal.
MIDI Mode This parameter allows you to define how the drawbar settings will respond to remote MIDI control change messages. Normally, you won’t need to change anything here. If you own a MIDI drawbar organ, however, you’ll probably want to use its hardware drawbars to control the EVB3. Most hardware drawbar organs utilize an independent control change number for every drawbar. Some models allow you to freely define these control change numbers.
The EVB3 Parameters You can open and close the EVB 3’s wooden lid by clicking the button underneath the Volume control. Click here to open the wooden lid Keep it open while reading this section of the manual, because it will cover every parameter in detail.
Drawbars The principles of additive synthesis with sine drawbars is further explained in “Additive Synthesis With Drawbars” on page 355. You can intuitively pick up the basic principles by playing a little with the drawbars. The further down you drag the drawbars, the louder the selected sine choir(s) will be—the drawbars behave like reversed mixer faders. MIDI control of the drawbars is also reversed, when using a standard MIDI fader box. Drawbars of the Upper and Lower manual, plus Pedal drawbars.
Volume Control and Expression Pedal The overall volume of the EVB3 is not only controlled by the software instrument channel’s volume fader and control change #7, but also with the Volume control in the EVB3 graphic user interface. Volume control Warning: The Volume must be lowered whenever crackling or other digital distortion occurs in the instrument channel. Volume levels over 0 dB can occur if you maximize all registers, play numerous notes, and make use of the Distortion effect.
Scanner Vibrato The vibrato of the organ itself must not be confused with the Leslie effect, which is based on rotating speaker horns. The EVB3 simulates both. The Scanner Vibrato is based on an analog delay line, consisting of several lowpass filters. The delay line is scanned by a multipole capacitor, which has a rotating pickup. It is a unique effect that cannot be simulated with simple LFOs. Like the Hammond B3, the EVB3 features three types of vibrato with different intensities (V1, V2, V3).
Percussion Percussion is only available for the Upper manual—same as the original B3. The percussion of an electro-mechanical organ is polyphonic, but is only (re)triggered after all keys have been released. If you release all keys, new notes or chords will sound with percussion. If you play legato, or sustain other notes on the Upper manual, no percussion will be audible. On the original B3, percussion is only available if the “B” preset key is selected (see “Preset Keys and Morphing” on page 333).
Preset Keys and Morphing The Hammond B3 is equipped with 12 switches, located below the lowest octave of both keyboard manuals. These are the preset keys, and are laid out as an inverted keyboard octave (black keys, white sharps). They are used to recall drawbar registrations. These presets could only be altered with a screwdriver on the original B3. Preset keys MIDI notes 24 to 35, the octave below the lowest octave of a (non-transposed) 5-octave keyboard, are used as the preset keys.
Cancel Key, Registering While Playing The lowest preset key (C) is the cancel key. If you depress it, all drawbars are moved to their minimum setting. The other 11 keys, from C# to B, recall registrations. You can edit recalled presets immediately. The preset memorizes these alterations instantaneously, with no further action required. This means that if you recall a new preset, the former preset memorizes the drawbar settings at the time the new preset was recalled.
Organ Parameters The Organ parameters adjust the overall behavior of your EVB3. Organ parameters The Lower Volume and Pedal Volume parameters are discussed in “Relative Volumes— Upper/Lower/Pedal” on page 329. The Perc parameter is discussed in “Percussion” on page 332. Max Wheels Calculating (emulating) all tone wheel generators consumes considerable CPU processing power. A reduction of this parameter value reduces the EVB3’s hunger for processing resources.
Bass Filter The sound of the Pedal drawbars often appears to be somewhat brilliant, within the overall musical context. To circumvent this issue, and to suppress the treble of the bass register, please make use of the Bass Filter. When active, you will only hear a solid bass organ fundamental in the bass register. Ultra Bass If you switch on Ultra Bass, another low octave will be added to the playable range of the Upper and Lower manuals.
Crosstalk The Hammond’s tone wheels are divided into compartments of four—with the same key, but in different octaves. There are two tonewheels, four octaves apart, on each rotating shaft. The signal of the Lower wheel contains a small amount of signal, induced by the higher wheel, and vice versa. This crosstalk can be adjusted with the Crosstalk slider. Note that crosstalk is only audible on certain wheels, avoiding rumble when chords are played.
Click On/Click Off These two knobs independently control the click volume for the beginning (Click On), and release, of the note (Click Off ). The click off is quieter, even if both controls are set to the same position. Click Min/Click Max Not only are the tone color and volume of clicks altered randomly, but also their duration. Click duration can vary between a short tick and a longer scratch. Minimum duration is defined by Click Min, and maximum duration with Click Max.
Note: The tones of clavinets, harpsichords, and pianos have inharmonicities in their harmonic structure. The frequencies of these overtones (harmonics) are not exact, whole-number multiples of the base frequency. They are only approximate and are, in fact, a little higher. This means that the overtones of lower (tuned) notes are more closely related to the main frequencies of the upper notes. Due to the lack of strings, this inharmonic relationship is not true of organs.
Sustain Synthesizer players call the time the note takes to fade out after the release of the key the release time. The EVB3 allows you to control this parameter as well; it’s called Sustain in the organ lexicon. The three controls allow for individual settings in the Upper (Up), Lower (Low) and Pedal (Ped) registers. Sustain parameters If you select Smart Mode, playing new notes will cut the sustain (release) phase of released notes.
You can suppress the brutal overtones of extreme distortions with two filters: Select Dist-EQ-Wah. Effect Bypass The Distortion, Wah, and EQ effects can be bypassed separately for the Pedal register. Set Effect Bypass to Pedal to do so. This avoids the entire bass portion of your organ being suppressed by the wah wah. It also avoids undesirable intermodulation artifacts, when utilizing the overdrive effect.
The reverb is always patched after the EQ, wah wah and distortion effects, but before the rotor effect. This means that the reverb always sounds as if it is played back through the rotor speaker. To hear the reverb after the rotor, switch off the organ’s reverb, and use an aux send to apply reverb to the instrument channel. Wah The name Wah Wah comes from the sound it produces. It has been a popular effect with electric guitarists since the days of Jimi Hendrix.
 Peak: In this mode, the wah wah will work as a peak (bell) filter. Frequencies inside the center frequency, which is controlled by the selected MIDI controller, will be emphasized.  CryB: This setting mimics the sound of the popular Cry Baby wah.  Morley 1: This setting mimics the sound of a popular wah pedal, manufactured by Morley. It features a slight peak characteristic.  Morley 2: This setting mimics the sound of the Morley distortion wah pedal. It has a constant Q.
Rotor Cabinet The Hammond story can’t be fully told without a chapter on the rotor sound cabinets, manufactured by Leslie. In fact, playing the B3 organ without a rotor cabinet is viewed as a special effect these days. The EVB3’s rotor cabinet section simulates not only the speaker cabinet itself, but also the microphones which pick up the sound. Rotor Speed Buttons These buttons switch the rotor speed as follows:  Chorale: Slow movement  Tremolo: Fast movement.  Brake: Stops the rotor.
Speed Control The Speed Control menu allows you to define controllers to switch the Rotor Speed buttons remotely. The ModWheel setting uses the modulation wheel to switch between all three speed settings. Brake is selected around the modulation wheel’s center position, while Choral is selected in the lower, and Tremolo in the upper third of the modulation wheel’s travel.
Acc/Dec Scale The Leslie motors need to physically accelerate and decelerate the speaker horns in the cabinets, and their power to do so is limited. Acc/Dec Scale determines the speed at which the motors can accelerate the rotors (time it takes to get the rotors up to a determined speed), and the length of time it takes for them to slow down. If the slider is set to its far left position, you can switch to the preset speed immediately.
Extended Parameters A number of additional parameters are accessible in the extended parameter view of the EVB3 window. Dry Level Adjusts the level of the dry signal, which can also be useful if the “Switches to dry sound” option is selected in the Brake menu, found below. Brake Menu The Brake menu offers two options that allow you to modify the EVB3’s Brake mode: Â Stops rotor: In this mode, the movement of the rotor is gradually slowed down to a total stop.
When using a two drawbar hardware controller, the “Drawbar affects” menu offers an additional mode that allows Hammond-like switching between two registrations. If you use the default setting (“Drawbar affects current preset key”), the drawbars will always change the registration of the currently active preset registration key. This works differently in a real” Hammond organ, where the drawbars only affect the Bb (upper manual) and B (lower manual) preset registrations.
MIDI Mode: RK This table describes the MIDI Control Change Message number assignment when MIDI Mode is set to RK. This is the correct setting if you use a Roland VK series or Korg CX-3 drawbar organ as a remote controller for the EVB3.
Controller Number MIDI Mode RK: Parameter Name Distortion 110 Distortion Type 111 Distortion Drive 112 Distortion Tone Click Levels 113 Click On Level 114 Click Off Level Balance 115 Main Volume 116 Lower Volume 117 Pedal Volume Rotor Fast Rate 118 350 Chapter 22 EVB3 Rotor Fast Rate
MIDI Mode: HS This tables describes the MIDI controller assignments when MIDI Mode is set to HS. This setting matches the controller mapping of Hammond XB-series organs. Switch MIDI Data Reduction off while recording data from an XB-Series organ (File > Project Settings > Recording).
MIDI Mode: NI This table describes the MIDI controller assignments when MIDI Mode is set to NI. This setting matches the controller mapping of the Native Instruments B4D controller.
Controller Number MIDI Mode NI: Parameter Name Distortion/Click 76 Distortion Drive 78 Distortion Tone 75 Click On Level Leslie Pan MSB Microphone Angle 3 Microphone Distance GP 8 Leslie Accelerate/Decelerate GP 7 Leslie Fast ModWheel MSB Leslie Speed 68 Controls Brake functionality: If Value = 0.0, switch Leslie to Brake. All other values switch Leslie to previous speed. MIDI Mode: NE This table describes the MIDI Control Change Message number assignment when MIDI Mode is set to NE.
Controller Number MIDI Mode NE: Parameter Name Chorus/Vibrato 85 Upper Vibrato on/off 86 Lower Vibrato on/off 84 Vibrato mode (selection goes from V1 to C3, C0 is excluded) Percussion 87 Percussion on/off 88 Percussion Volume (soft/normal) and Time (short/long) 95 Percussion Harmonic (2nd/3rd) Equalizer 113 EQ High 114 EQ Low Distortion/Click 111 Distortion Drive Leslie GP 6 354 On/off GP 7 Leslie Speed GP 8 Controls Brake functionality Chapter 22 EVB3
Additive Synthesis With Drawbars The Hammond B3 is the classic drawbar organ. As with an acoustic pipe organ, the registers (drawbars, or stops on a pipe organ) can be pulled out, in order to engage them. But in contrast to a pipe organ, the B3 allows seamless mixing of any drawbar registers. The more you drag the drawbars down, the louder they will become.
Note: 2 2/3' is the fifth over 4'. 1 3/5', is the major third over 2'. 1 1/3' is the fifth over 2'. In the bass range, this can lead to inharmonic tones, especially when playing bass lines in a minor key. This is because mixing 2', 1 3/5' and 1 1/3' results in a major chord. Residual Effect The residual effect is a psychoacoustic phenomenon. Human beings can perceive the pitch of a note, even when the fundamental is completely missing.
A Short Hammond Organ Story Three inventions inspired Laurens Hammond (1895–1973), a manufacturer of electric clocks, to construct and market a compact electro-mechanical organ with tone wheel sound generation. The Telharmonium by Thaddeus Cahill was the musical inspiration, Henry Ford’s mass production methods, and the domestic synchron clock motor were the other factors. The Telharmonium was the first musical instrument that made use of electromechanical sound generation techniques.
Tonewheel Sound Generation Tonewheel sound generation resembles that of a siren. Of course, there’s no air being blown through the holes of a revolving wheel. Rather, an electro-magnetic pickup, much like a guitar pickup is used. A notched metal wheel, called a tone wheel, revolves at the end of a magnetized rod. The teeth of the wheel cause variations in the magnetic field, inducing an electrical voltage.
23 EVD6 23 The EVD6 is a virtual emulation of the classic Hohner Clavinet D6. The sound of the Hohner Clavinet D6 is synonymous with funk, but was also popularized in the rock, pop, and electric jazz of the 1970s, by artists like Stevie Wonder, Herbie Hancock, Keith Emerson, Foreigner, and the Commodores.
The EVD6 Parameters Most of the slider parameters on the EVD6 interface are mapped to zero-centered ranges—if a slider is in its middle neutral position, it doesn’t affect the base sound of the selected EVD6-model. If the slider is moved to the left or right, it will scale the original parameter value positively or negatively by that amount.
There are two monophonic settings: mono and legato. Each setting provides only one voice for playing the EVD6. In the mono setting, the EVD6 voice is triggered each time a key is pressed. In the legato mode, the EVD6 sound shaping processes is not triggered if the notes are played legato—only the pitch changes. If the notes are played staccato, an EVD6 voice with all sound shaping processes is triggered. Minimal CPU power is used when the instrument is operated monophonically.
Note: When applying Warmth and Stretch, you should consider that these parameters may result in a detuned sound, which is similar to a heavy chorus effect. Pressure On an original D6, applying pressure (aftertouch) to a depressed key raises the pitch slightly. The Pressure parameter allows you to do this, or alternately lower the pitch by pressure. Click-hold, and use your mouse as a slider to adjust. Range: –1.00 to +1.
C/D Switch A/B Switch What It Does Down Down Neck pickup—warm sound Down Up Bridge pickup—bright sound Up Up Both pickups—full sound Up Down Both pickups out of phase—thin sound Stereo Spread The Stereo Spread control offers two sections: Pickup and Key. You can use both spread types at the same time. They will automatically be mixed. The area around the circular Stereo Spread button graphically displays the effect of both stereo spread parameters.
Model The Model parameter allows you to select a basic type of tone, or model. Each model has its own unique tonal characteristic designed to create very different sounds. Each model is an instrument in its own right, and can immediately be played, without any further editing. Each model is explained below. To select a model, click-hold in the area between the Stereo Spread and Level controls, and make your choice from the pop-up list. Release the mouse button once your selection is made.
Basic Basic, simple clavinet. Domin(ation) A powerful model with a strong and punchy attack—reacts more aggressively to velocity than other models. GuruFnk (Guru Funk) In the lower bass-octave ranges, the string oscillations become increasingly resonant over time, until they finally collapse (after 20 to 30 seconds). Higher notes have a much shorter decay, which also applies to their resonating behavior. This model invites heavy, funk-style bass playing in the lower octaves.
Special Notes About the Models You may note some spots on the keyboard where the sound changes significantly between adjacent keys. This is intentional, and reflects the behavior of some of the real clavinet models emulated by the EVD6. The original D6 has some strong key-to-key timbral differences, with the most obvious one being between the highest, wound string, and the lowest, non-wound string.
The Damper Wheel position is saved with the sound. The Damper Ctrl (number) parameter allows you to select the MIDI controller that moves the Damper Wheel. Click and choose a controller number/name selection from the menu. Release the mouse button, once your selection is made. Note: You can use MIDI Velocity to control the Damper Wheel. Just select the Velocity parameter from the menu. When you move the software-wheel via MIDI, it moves on-screen as well.
Shape Adjusts the attack shape, allowing you to simulate the hardness of the rubber hammers in the original D6. As the instrument aged, the hammers would become worn, split, and so on, which had an impact on the overall brightness/tone of the D6. Negative values (to the left) provide a softer attack, while positive values result in a harder attack. Range: –1.00 to +1.00 Brilliance Controls the harmonic content of string excitation. Positive values (to the right) result in a sharper sound.
KeyOn/KeyOff Button Press the appropriate button to select the type of velocity information that should be used for release click level modulation—click the KeyOn button, if you wish to use your attack velocity (how hard you hit the keyboard) as the value for the key click. If you wish to use your release velocity (how quickly you release the keys on your keyboard) to determine the value of the key click, click the KeyOff button. Note: KeyOff requires a keyboard with release velocity facilities.
Damping Lets you modify the damping of strings. Damping is essentially a faster decay for the higher partials/harmonics in a sound, and is a property of the string material used (high damping for catgut strings, medium damping for nylon strings, low damping for steel strings). Sonically, damping results in a more mellow and rounded, or woody sound, dependent on the Model in use.
Pickup Parameters The original D6 is equipped with two electromagnetic pickups, much like those found in electric guitars: one below the strings (lower) and one above (upper). Pickup Position In contrast to the fixed pickups of the original instrument, the EVD6 pickups can be set to arbitrary positions and angles. To do so, simply click-hold on one end of the desired pickup (Upper or Lower) and drag the end to another position. Release the mouse button when done. Both values can be moved simultaneously.
Pickup Mode Clicking the AB and CD switches changes the virtual wiring of the two pickups. The current wiring, the EVD6 calls it Pickup Mode, is displayed in the Pickup Mode panel. You can also click directly on the Pickup Mode panel, and choose the desired mode from a menu.  C + A = Lower  C + B = Upper  D + A = Lower-Upper  D + B = Lower+Upper Also see “Stereo Spread” on page 363, and “Pickup Switches” on page 362.
Compressor The compressor always precedes the Distortion effect. This allows you to increase/ decrease the perceived gain, to provide the desired input level to the Distortion circuit. The compressor allows you to create really crunchy distortions, coupled with wah, or phaser. It can also be useful for enhancing the keyclick sound, and emphasizing harmonics in the various models. Compression Ratio The Compression Ratio panel allows you to adjust the slope of the compression applied.
Range Determines the cutoff frequency of the filter set with Wah Mode. With Range set to the left, the cutoff will only move in a narrow range. To provide a wider control range, turn the Range knob to the right. Envelope (Depth) Sets the sensitivity of the envelope in respect to your performance and thus the resulting filter modulation depth. The envelope shape follows the dynamics of your performance. An auto wah effect is produced by using an envelope follower to control the filter cutoff automatically.
Modulation The EVD6 features a Modulation unit with three switchable modulation effect types. Modulation unit Mode The Mode panel allows you to select either a Phaser, Flanger, or Chorus as the modulation effect. Click-hold, and make your choice from the pop-up menu. Phaser The Rate parameter adjusts the speed of phasing, and the Intensity parameter adjusts the depth of phasing. Ranges: Rate 0.
FX Order The order of the serial effects combination can be selected here. The four choices are: Â Â Â Â WDM: DWM: MDW: WMD: Wah > Distortion > Modulation Distortion > Wah > Modulation Modulation > Distortion > Wah Wah > Modulation > Distortion Just like the foot pedals which could be freely connected to each other in series, the EVD6 effect section invites you to experiment.
A Brief History of the Clavinet German Company, Hohner, manufacturer of the Clavinet, were known mainly for their reed instruments (harmonicas, accordions, melodicas, and so on), but had made several classic keyboards, prior to the first incarnation of the Clavinet, known as the Cembalet. Musician and inventor, Ernst Zacharias, designed the Cembalet in the 1950s. This was intended to be a portable, amplifiable version of the Cembalo, or Harpsichord.
How the D6 Clavinet Works Each D6 keyboard key forms a single arm lever. When a key is depressed, a plunger underneath touches the string and presses it onto an anvil. The string impinges on the anvil with a strength determined by key velocity. This affects the dynamics of the sounding string. These mechanical vibrations are converted into electrical frequencies through magnetic pick-ups which are amplified and reproduced through the loudspeaker.
24 EVP88 24 The EVP88 virtual electric piano instrument simulates the sound of different Rhodes and Wurlitzer pianos, as well as the sound of the Hohner Electra piano. The sounds of various Fender Rhodes pianos are among the most popular keyboard instrument sounds used in the second half of the twentieth century. The various Rhodes models have been popularized in a wide range of musical styles, ranging from pop, rock, jazz, and soul, as well as more recent genres like house and hip hop.
The EVP88 Parameters This section outlines the various controls available on the EVP88 front panel. Global Parameters Global parameters affect the entire EVP88 instrument, rather than specific electric piano models. Model The large Model dial allows you to choose the electric piano model. When selecting a new model, all currently sounding voices are muted, and all parameters are reset to standard values.
Tune The global Tune setting lets you tune the EVP88 in one cent increments. A value of 0 equals concert-pitch A 440 Hz. The range is ±50 cents or, in more music-related terms, plus/minus half a semitone. For transpositions in semitone or octave steps, please use the Region Parameter box in the Arrange window, as per any standard MIDI instrument. Modeling Parameters The modeling parameters specifically affect the currently selected model. Decay Decay time of the piano sound.
Note: Using Logic Studio’s plug-ins, you can process the upper notes differently to the lower ones. With appropriate signal processing routings, you can, for example, add some bass via an EQ in the left bass channel and apply a little echo to the higher notes. Be creative! Stretch and Warmth The EVP88 is tuned to an equal-tempered scale. However, you can deviate from this standard tuning, and stretch the tuning in the bass and treble ranges, much like acoustic pianos (especially upright pianos).
Effects The EVP88 features built-in Equalizer, (Over)Drive, Phaser, Tremolo, and Chorus effects. Equalizer The Equalizer allows you to boost or cut the high and low frequency ranges of the EVP88 sound. Treble This is a conventional filter for the high frequency range. Depending on the model selected, shelving or peak type filters are utilized, with optimized frequency ranges for each model pre-selected. Bass This is a conventional filter for the low frequency range.
Tone The Tone control equalizes the sound before the virtual tube amplifier circuit amplifies or distorts it. You can choose a more mellow tonal color here, and still boost the treble with the equalizer after the overdrive circuit. If you prefer harsh distortion characteristics typical of overdriven transistor stages, use higher tone parameter values. If the sound becomes too hard, you can defeat the treble via the Treble control, post the overdrive process.
Stereophase Relative phase shift between the left and right channels, ranging from 0° to 180°. With 0° selected, the effect is most intense, but not stereophonic. With 180° selected, the effect symmetrically rises in the left channel while simultaneously falling in the right channel, and vice versa. Tremolo A periodic modulation of the amplitude (level) of the sound is known as tremolo. The modulation is controlled via an LFO.
Chorus The well-known chorus effect is based on a delay circuit, the delay time of which is permanently modulated by an LFO, while the delayed effect signal is mixed with the original dry signal. It is the most popularly used effect on electric piano sounds. The single parameter regulates the intensity (the amount of delay time deviation), while the LFO rate is fixed at 0.7 Hz. Pay close attention when using high values as this may result in the piano sounding detuned.
Emulated Electric Piano Models This section provides some background information on the instruments emulated by the EVP88. Rhodes Harold Rhodes (born 1910) constructed arguably the most commonly known and widely used electric piano. Designed in 1946 as a piano surrogate for practice, education, and army entertainment, guitar manufacturer Fender successfully marketed the Rhodes piano beginning in1956. The Fender Rhodes has become one of the most popular musical instruments in jazz, especially electric jazz.
The Rhodes piano was also made available as a suitcase piano (with pre-amp and twochannel combo amplifier) and as a stage piano, without amplifier. Both of these 73-key portable versions have a vinyl-covered wooden frame and a plastic top. In 1973, an 88 key model was introduced. Smaller Celeste and bass versions were less popular. The Mk II (1978) had a flat top instead of a rounded one. This allowed keyboardists to place extra keyboards on top of the Rhodes.
Wurlitzer Piano This well-known manufacturer of music boxes and organs also built electric pianos which helped write pop and rock music history. The 200 series Wurlitzer pianos are smaller and lighter than the Rhodes pianos, with a keyboard range of 64 keys from A to C and an integrated amplifier and speakers. The action resembles that of a conventional acoustic piano. It can be played with velocity sensitivity, just like the Rhodes.
EVP88 and MIDI This section outlines some of the special MIDI control features available to the EVP88. Adaptation of Your MIDI Keyboards Velocity Sensitivity The EVP88 responds with extreme sensitivity to the velocity information transmitted with MIDI note messages. It’s advisable to set the velocity and dynamic track parameters in Logic Pro with care. You can try the following tip to fine-tune the velocity curve if you find that you’re not getting the right feel with your MIDI keyboard.
25 EXS24 mkII 25 The EXS24 mkII is a software sampler. This means that instead of having a built-in sound set, it plays back audio files (called samples) that you load into it. These samples are combined into tuned, organized collections of samples called sampler instruments. You can use the EXS24 mkII to play, edit, and create sampler instruments. You can assign the samples (in sampler instruments) to particular key and velocity ranges, and process them with the EXS24 mkII filters and modulators.
 Instrument Editor: Used to create and edit sampler instruments. Using the EXS24 mkII typically involves the following steps: 1 Load or import a sampler instrument. 2 Change the overall sound of the sampler instrument in the EXS24 mkII Parameters window by turning the knobs, pressing switches, and moving sliders. You can also automate these controls, enabling dynamic changes over time. 3 Edit specific samples in the Instrument Editor.
Sampler instruments are distinct from plug-in settings, which are loaded and saved in the plug-in header. Plug-in settings reside above the sampler instruments in the file hierarchy: A setting contains a pointer to a sampler instrument, and when a new setting is selected, the sampler instrument it points to is automatically loaded.
Loading Sampler Instruments The EXS24 mkII ships with a ready-to-play sampler instrument library. To load an instrument: 1 Click the Sampler instrument field directly above the Cutoff knob in the EXS24 mkII Parameters window. This opens the Sampler Instruments menu. 2 Browse to, and select, the desired sampler instrument.
To browse to the next or previous instrument of your sampler instrument library, do one of the following: m Click the plus or minus button to the left and right of the Sampler Instruments menu. m Choose Next Instrument or Previous Instrument in the Sampler Instruments menu (or use the Next EXS Instrument and Previous EXS Instrument key commands).
It is strongly recommend that you copy any EXS sampler instruments, along with their associated audio files, to your hard drives. This way, you always have direct, immediate access to your sampler instruments without searching for and inserting CD-ROMs or DVDs. This also allows you to organize your sampler instruments in accordance with your needs. To copy sampler instruments to your hard drives: 1 Copy the sampler instrument file into the ~/Library/Application Support/Logic/Sampler Instruments folder.
Managing Sampler Instruments As your sample library grows, the list of sampler instruments will also expand. To aid you in keeping the list of sampler instruments manageable, the EXS24 mkII features a simple, but sophisticated file management method. To organize your sampler instruments into a preferred hierarchy: 1 Create a folder in the Finder—Basses for example—and drag it into the desired Sampler Instruments folder. 2 Drag the desired EXS24 mkII sampler instruments into this newly created folder.
∏ Tip: You can also do this by setting up your project to copy EXS24 sampler instruments and samples into the project folder. For more information, see the Logic Pro 8 User Manual. Searching for Sampler Instruments In order to minimize the number of sampler instruments displayed in the Sampler Instruments menu, you can make use of the Find function. This will limit the Sampler Instruments menu to only display sampler instrument names that contain the search word.
Importing Sampler Instruments The EXS24 mkII is compatible with the AKAI S1000 and S3000, SampleCell, ReCycle, Gigasampler, DLS, and SoundFont2 sample formats, as well as the Vienna Library. Importing SoundFont2, SampleCell, DLS, and Gigasampler Files The EXS24 mkII automatically recognizes SoundFont2, SampleCell, DLS, and Gigasampler files placed inside the Sampler Instruments folder and converts them into sampler instruments.
The procedure outlined above can also be used to import SoundFont2 and SampleCell Bank files, which contain multiple sounds, in addition to single instrument files. If you load a SoundFont2 or SampleCell Bank into the EXS24 mkII, it creates a Bank and a Samples folder, named after the SoundFont2/SampleCell Bank file. The word Bank or Samples is appended to each folder name. All sounds contained in the bank will automatically have an EXS sampler instrument file created, and placed into the new Bank folder.
Note: You can store your imported sampler instruments in any folder on any of your computer’s hard drives. To make sure that these instruments are displayed in the Sampler Instruments menu, you must create an alias pointing to this folder within the ~/Library/Application Support/Logic/Sampler Instruments folder.
4 Click OK. The EXS24 mkII generates a zone for each slice of the imported ReCycle file, and assigns these zones to one group (see “Editing Zones and Groups” for more information on zones and groups). The new EXS instrument will be assigned the name of the ReCycle loop. Should an EXS instrument of that name already exist, a # sign and a number will be appended.
Generating a MIDI Region From a ReCycle Instrument You can generate a MIDI region from imported ReCycle files in EXS instruments, triggering the imported slices at the timing defined by the ReCycle files. To generate a new MIDI region from a ReCycle instrument: m Choose Instrument > ReCycle Convert > Extract Region(s) from ReCycle Instrument. The MIDI regions are created on the currently selected track, at the current project position (rounded to bars).
2 Insert an AKAI format sample disc into your CD-ROM drive. The display updates to show the contents of the CD-ROM. The Partition column will display information, with Partition A, Partition B (and so on) entries listed. 3 To view the contents of the Partitions, click once on the appropriate entry with the mouse button. This will display the Volume information contained within the Partition.
The selected Partition, Volume, or Program will be imported, along with all associated audio files. Logic (folder) Sampler Instruments (folder) AKAI Samples (folder) AKAI sampler instruments Audio files (samples) Â Any audio files imported will be stored within a folder which matches the name of the Volume. This folder is created within the ~/Library/Application Support/Logic/ AKAI Samples folder. Â The sampler instruments created by the import procedure match the Program names.
Additional AKAI Convert Parameters Within the AKAI Convert window, you will find the additional parameters listed below. Save converted instrument file(s) into sub folder You can use this when importing an entire CD. This creates a folder name which reflects the CD-ROM’s name. Alternately, you may want save your converted instruments into a sub-folder based on a category, such as Strings.
The same is true for drum CD-ROMs where single programs contain one instrument from a complete drum kit (kick, snare, hi-hat, and so on as separate entities) You’ll probably want these single AKAI programs to be merged into a single EXS sampler instrument—as a full drum kit. There are, however, a number of AKAI CD-ROMs where a single program of an AKAI Volume contains the entire instrument, and where other programs in the same Volume have the same MIDI channel and MIDI program change number preset.
Parameters Window The EXS24 mkII Parameters window contains settings that determine how the EXS24 mkII processes the entire loaded sampler instrument.
General Parameters The following section describes the general parameters of the EXS24 mkII. Legato/Mono/Poly Buttons These buttons determine the number of voices used by the EXS24 mkII (in other words, how many notes can be played simultaneously): Â When Poly is selected, the maximum number of voices is set via the numeric field alongside the Poly button. To change the value, drag up or down to increase or decrease polyphony.
The voices are equally distributed in the panorama field and are symmetrically detuned, dependent on the Random knob value. Note: The number of voices actually used per note increases with the number of layered sample zones. Sampler Instruments Menu You can click in the Sampler Instruments menu to load a sampler instrument into the EXS24 mkII. See “Loading Sampler Instruments” for complete information on the Sampler Instruments menu.
 (Recall default EXS24 mkI settings): For sampler instruments that were created using the older EXS24. This will recall the parameter settings for the former version of the selected instrument, especially the former modulation paths (see “EXS24 mkI Modulation Paths” on page 423). For sampler instruments that were created in the EXS24 mkII, this parameter is of little use.  Extract MIDI Region(s) from ReCycle Instrument: Allows you to extract the regions contained in a ReCycle instrument.
Crossfade (Xfade) parameters If you are familiar with the concept of layering sample zones by Velocity Range, the Xfade parameters allow you to crossfade between layered sample zones with adjacent Velocity Range settings. If you are not familiar with this idea, here’s a short explanation: When assigning a sample to a zone, you can set the lowest and highest velocity that will trigger that zone. The range in between these values is the zone’s Velocity Range.
In the Type menu, you can choose between three different choices of fade curves for the velocity crossfade: Â dB lin (dB linear): A logarithmic curve, so that the crossfade sounds even on both sides. Â linear (gain linear): A convex crossfade curve, so that the crossfade sounds as if nothing is happening at first, but then with a rapid volume fade towards the end. Â Eq. Pow (equal power): A nonlinear curve in which the level rises faster in the beginning but rises back to normal more slowly.
Pitch Bend Up This parameter determines the upper limit of pitch bending (in semitones) that can be introduced by moving the pitch bend wheel to its maximum position using this parameter. Pitch Bend Up has a range of 0 semitones (the pitch bend wheel in its maximum position will not increase the pitch at all) to 12 semitones (the pitch bend wheel in its maximum position will increase the pitch a full octave).
Glide The effect of this slider depends on the setting of the Pitcher slider: When Pitcher is centered, Glide determines the time it takes for the pitch to slide from one note to another (called portamento). When the Pitcher parameter is set to a value above its centered value, Glide determines the time it takes for the pitch to glide down from this higher value back to its normal value.
When both halves of the pitcher slider are set below or above the centered position, either a low or high velocity will slide up/down to the original pitch. Dependent on the position of the upper/lower halves of the slider in relation to the center position, the time required for the slide up/down to the original note pitch can be adjusted independently for both soft/hard velocities. Filter Parameters These parameters control the EXS24 mkII filter section.
Highpass (HP) Click the button under the HP label to engage the highpass filter. The highpass filter is a 2 pole (12 dB/Oct.) design. A Highpass filter reduces the level of frequencies that fall below the cutoff frequency. It is useful for situations where you would like to suppress the bass and bass drum in a sample, for example, or for creating classic highpass filter sweeps. Bandpass (BP) Click the button under the BP label to engage the bandpass filter. The bandpass filter is a 2 pole (12 dB/Oct.
Key This knob defines the amount of filter cutoff frequency as determined by note number. When Key is fully turned to the left, the cutoff frequency is not affected by the note number, and is identical for all notes played. When Key is set fully right, the cutoff frequency follows the note number 1:1—if you play one octave higher, Cutoff is also shifted by one octave. This parameter is very useful in avoiding overly filtered high notes.
Volume The Volume knob is the main volume parameter for the EXS24. Move this knob to find the right balance between avoiding distortion and getting the best (highest) resolution in the channel fader and the Level via Vel slider. Key Scale This parameter modulates the sound’s level by note number (position on the keyboard). Negative values increase the level of lower notes. Positive values increase the level of higher notes.
LFO Parameters The EXS24 mkII includes three LFOs (low frequency oscillators) which can be used as modulation sources. This section explains their parameters. LFO 1 EG This knob allows LFO 1 to be faded out (when the knob is pointing inside the Decay area) or faded in (when the knob is pointing in the Delay area). In the centered position (which can be set by clicking on the small 0 button), the LFO intensity is constant. LFO 1 Rate This is the frequency of LFO 1.
LFO 2 Rate The frequency rate of LFO 2 can be set in note values (left area), or in Hertz (right area). In the centered position (which can be set by clicking on the small 0 button), the LFO is halted, and generates a constant modulation value at full level (DC = Direct Current). LFO 3 Rate There is a third LFO available—which always uses a triangular waveform. LFO 3 can oscillate freely between 0 and 35 Hz, or can be tempo synchronized in values between 32 bars and 1/128 triplets.
3 Set the modulation depth with the green triangular fader on the right side of each modulation path. In the example above, the LFO 1 speed is modulated by channel pressure (aftertouch) messages of a MIDI keyboard. You have the option of inserting another modulation source in the middle slot labeled via. The via modulation source doesn’t directly modulate the destination, but instead it modulates the source—in essence, modulating the modulator.
The following example shows an inverted via modulation source. You can see how the green and orange triangles have swapped positions. The orange triangle always marks the modulation depth for the maximum value of the via source, while the green triangle always marks the modulation depth if the via source is at its minimum value. They are reversed by inverting the modulation.
    LFO 2 to Filter Cutoff via ModWheel Velocity to Filter Cutoff Envelope 1 to Filter Cutoff via Velocity LFO 2 to Pan via ModWheel You can, of course, freely alter the settings of these modulation paths. To exchange modulation sources with sources that were not available in EXS24 mkI, for example (see the complete list of sources and destinations below). Note: For technical reasons, the settings of the modulation matrix can not translate backwards to the EXS24 mkI.
 Time  Hold Note: Controllers 7 and 10 are marked as (not available). Logic Pro uses these controllers for volume and pan automation for the audio channel strips. Controller 11 is marked as (Expression). It has a fixed connection to this functionality, but it can also be used to control other modulation sources. Sample Select This modulation destination deserves a more thorough explanation. By default, Sample Select is controlled by velocity through the default Velocity to Sample Select modulation path.
To open the Instrument Editor: m Click the Edit button in the upper right corner of the EXS24 mkII Parameters window (or use the Open EXS24 Instrument Editor key command). The Instrument Editor features two views: Zones view and Groups view. Zones view displays zones and their parameters, in the parameter area. In Zones view, the area above the keyboard becomes the Zones area, and displays bars for the created zones. Groups view shows groups and their parameters.
 Parameters area: Displays the parameters of the zone group chosen in the Zones column.  Velocity area: Shows the velocity range of the selected zone.  Zones/Groups area: Shows the zones or groups graphically placed above the keyboard.  Keyboard: Use to trigger notes for the EXS24 mkII in the currently selected track. The keyboard also serves as a visual reference for the placement of the zones or groups in the Zones or Groups area.
Zones A zone is a location into which a single sample (or audio file, if you prefer this term) can be loaded. The sample loaded into the zone is memory resident—it uses the RAM of your computer. A zone offers various parameters for controlling the playback of the sample. Each zone allows you to determine the range of notes over which the sample should be heard (Key Range), and the root key—the note at which the sample sounds at its original pitch.
The “Preview audio file in EXS instrument” option temporarily replaces the sample files in the currently selected zone. The zone is not directly triggered by activating this option, but it can be triggered via MIDI notes while the file selection dialog is open and different files are being selected. The selected sample can be heard as part of the zone, inclusive of all synthesizer processing (filters, modulation, and so on). 5 Once you find a sample you’d like to use, click Open to load it.
To load multiple samples in one operation: 1 Choose Zone > Load Multiple Samples in the Instrument Editor (or use the Load Multiple Samples key command). 2 Browse to the desired location, then use the Add, Add All, Remove, or Remove All buttons to select the desired samples. 3 Click the Done button when you are finished.
Groups Imagine a drum kit has been created, with a number of different samples being used in several zones, mapped across the keyboard. In many musical circumstances, you might want to adjust the sound editing parameters of each of the samples independently— to alter the decay of the snare, or to use a different cutoff setting for the hi-hat samples, for example. This scenario is where the EXS24 mkII’s groups feature comes in. Groups allow for very flexible organization of samples.
m Drag a zone (or multiple selection of zones) out of one group to Ungrouped Zones to change the group assignment to unassigned. m Drag an ungrouped zone (or multiple selection of zones) into the empty area below Ungrouped Zones to create a new group containing the dragged zone (or zones). m Drag a zone (or multiple selection of zones) out of one group folder to the empty area below Ungrouped Zones to create a new group containing the dragged zone (or zones).
 Toggle Selection (also available as key command): Toggles the selection between the currently selected zones or groups and all currently unselected zones or groups. You can also click on zones and groups in the Parameters area:  Clicking on the parameters of a single zone or group selects that zone or group.  Shift-clicking multiple zones in the Zones view selects all the zones between the two clicked zones.  Command-clicking multiple zones selects each clicked zone.
∏ Tip: Double-clicking the Audio File column—when an audio file is loaded—opens the audio file in the Sample Editor. When no audio file is loaded, the audio file selector opens. Group Shows the group assignment of a zone. See “Groups” for more information. Pitch Parameters Key allows you to determine the root note of the sample—in other words, the note at which the sample will sound at its original pitch. The Coarse and Fine fields allow coarse and fine-tuning of the sample in semitone and cent increments.
The Routing parameter determines the outputs used by the zone. Choices include the main outputs, and paired channels 3 and 4, 5 and 6, 7 and 8, 9 and 10, or individual outputs 11 through to 16. This allows individual zones to be routed independently to aux channels in a multi-output EXS24 mkII instance. Playback Parameters To enable the sample to change pitch when the sample is triggered by different keys, switch on the Pitch checkbox.
 Xfade (Crossfade): In a crossfaded loop, there is no hard cut between the loop end and loop start points. Rather, the loop end and start points are crossfaded for a smooth transition. This is especially convenient with samples that are hard to loop, and would normally produce clicks at the transition point—the join in the loop. The Crossfade field allows you to determine the time of the crossfade.
The Instrument Editor does not offer a graphic representation of a sample’s waveform. Because of this, the only way to set the start and end points or loop points of a sample within the Instrument Editor is numerically. Thankfully Control-clicking the loop start and end point parameters will open a shortcut menu, allowing you to open the selected sample in the Logic Pro Sample Editor (or the external sample editor set in the preferences).
To use the Sample Editor Loop commands: 1 Choose either of the selection commands in the Edit menu of the Sample Editor: Â Sample Loop → Selection: The loop area (defined by the Loop Start and End points) is used to select a portion of the overall audio file. Â Selection → Sample Loop: The selected area is used to set the Loop Start and End points. 2 Once you’ve selected the desired area using either of the commands above, choose Edit > Write Sample Loop to Audio File.
Velocity Range The two velocity range parameters allow you to set up a velocity range for the group. Use these parameters for sounds where you wish to mix, or switch between, samples dynamically by playing your MIDI keyboard harder or softer—with layered sounds, or when switching between different percussion samples, for example. Low sets the lowest velocity at which the group will sound, and High sets the highest velocity at which the group will sound.
Filter Offsets Allow you to offset the Cutoff and Resonance settings of the Parameters window separately for each group. This can be useful if you want the initial impact of a note to be unfiltered for one group, but not others. Envelope 1 and Envelope 2 Offsets Parameter Use these parameters to offset the envelope settings from the Parameters window separately for each group.
Click the plus icon in the upper right corner of the Select Group By column to refine the group selection conditions. Click the minus icon to remove a Select Group By condition, and broaden the group selection criteria.
2 Click-hold and drag the zone or group to the desired position. Note: You can also change the root key of a zone, by pressing Option and Command simultaneously, while dragging the zone. You can move multiple zones or groups at the same time by rubber-band, Shift-click, or Command-click selecting them. To change the start or end note of a zone or group: 1 Move the mouse cursor to the beginning or end of a zone or group. The cursor will change to a two header arrow.
To edit the velocity range of a zone or group: 1 Click the Show Velocity button at the top right of the Instrument Editor (or use the Show/Hide Velocity key command). The Velocity Display area will open above the Zones or Views Display area. 2 Click on one or more zones or groups in the Display area. The velocity bars of the selected zones will be highlighted in the Velocity Display area. 3 Click-hold on either the High or Low value of the velocity bar of the zone or group you wish to edit.
Sorting Zones and Groups You can easily sort zones and groups in the EXS24 mkII Instrument Editor by clicking the sub-column heading that you wish to sort by. For example, if you want to sort your zones by name, click the Name sub-column heading under the Name column, and your zones will be sorted alphabetically by name.
Saving, Deleting, Renaming, and Exporting Instruments You can access all basic sampler instrument operations in the Instrument Editor’s Instrument menu. Save Saves the currently loaded sampler instrument. When you create a new instrument and save it for the first time, you will be asked to give it a name. If you have edited an existing sampler instrument and save it via this command, the existing file name is used and the old instrument is overwritten. You can also use the Save Instrument key command.
Setting Sampler Preferences The EXS24 mkII offers a separate Sampler Preferences window, allowing you to configure various operational and sample related preferences, such as sample rate conversion quality, velocity responsiveness, sample storage, search-related parameters, and so on. To open the Sampler Preferences window, do one of the following: m Click the Options button in the Parameters window, then choose Preferences in the pop-up menu. m Choose Edit > Preferences in the Instrument Editor.
Search Samples On Menu Determines the location that instruments samples should be searched in. You may either choose the drives normally used by the operating system or external SCSI, FireWire, or USB drives, accessible directly or over a network. Drives can be selected individually, or grouped as follows: Â Local Volumes internal storage media (hard disks and CD ROM mechanisms) attached to or installed in the computer directly. Â External Volumes storage media accessible over a network.
Note: There may be cases where a sound designer has used multiple numbers in a filename, which is common with loops, with one value being used to indicate tempo— “loop60-100.wav”, for example. In this situation, it isn’t clear which, if either of the numbers, indicates a root key or something else: 60 or 100 could indicate the file number in a collection, tempo, root key, and so on. You can set a value of 8 to read the root key at position (letter/character) eight of the filename—namely the 100 (E6).
Configuring Virtual Memory These days, many sample libraries contain many gigabytes of audio samples in order to create the most accurate sampler instruments possible. Often, these gigantic sample libraries are too large to fit into your computer hardware’s random access memory (RAM) all at once. To let you use these huge sample libraries, the EXS24 mkII can use a portion of your hard drive as virtual memory.
 Performance section: Shows the current disk I/O traffic and the data not read from disk in time. If these numbers start rising, the EXS24 mkII may glitch when trying to stream your samples from the disk in time with your performance. If you notice these values rising to high levels, you should change the general settings to free up additional RAM for virtual memory use.
26 External Instrument 26 You can use the External Instrument to route your external MIDI sound generators through the Logic Pro Mixer, allowing you to process them with Logic Pro effects. Ideally, you will use a multi input and output audio interface, to avoid constant repatching of devices. The External Instrument can be inserted in software instrument channels in place of a software instrument.
Using the External Instrument The following section outlines the steps required to route external MIDI sound generators through the Logic Pro Mixer. To process external MIDI instruments with effects: 1 Connect the output (or output pair) of your MIDI module with an input (pair) on your audio interface. Note: These can be either analog or digital connections if your audio interface and effects unit are equipped with either, or both. 2 Create an instrument channel.
27 Klopfgeist 27 Klopfgeist is an instrument that is optimized to provide a metronome click in Logic Pro. Klopfgeist is inserted on instrument channel 128 by default, and used to generate the MIDI metronome click. Theoretically, any other Logic Pro or third-party instrument could be used as a metronome sound source on instrument channel 128. Similarly, Klopfgeist can be inserted on any other instrument channel for use as an instrument.
 Level Via Vel slider and fields: Determine the velocity sensitivity of Klopfgeist. The upper half of the two-part slider determines the volume for maximum velocity, the lower half for minimum velocity. By clicking and dragging in the area between the two slider segments, you can move both simultaneously.
28 Sculpture 28 Sculpture is a synthesizer that generates sounds—based on a simulated string or bar—that is in motion, or vibrating. To keep things clear, this chapter will always refer to the string, even though many of the sounds you can create with Sculpture have nothing in common with what you’d expect from a stringed instrument! Sculpture uses a method of synthesis called component modeling.
The Synthesis Core of Sculpture This section is designed to give you a feel for the way Sculpture works. Component modeling, as you’ll discover, is quite different to traditional synthesis methods—and so are the results! The layout of parameter descriptions follows the signal path (shown in the diagram) of the core synthesis engine. Please check out the options of each parameter as you read about them. This will give you a better feel for where things are, and what’s available.
Up to three objects of different types are used to excite or disturb the vibration of the string. These objects can be positioned anywhere along the string, and offer multiple parameters for adjustments to their properties. The string itself doesn’t make a sound unless it is stimulated (excited or disturbed) by the objects.
Sculpture goes far beyond the mere creation of an infinite number of base timbres, however. One of the key differences between Sculpture’s string and a traditional synthesizer’s waveform is that the base timbre (provided by the string) is in a constant state of flux. Put another way, if Sculpture’s string is still vibrating for a specific note, retriggering that same note will interact with the ongoing vibration.
Keep the Sculpture flowchart diagram (on page 456) handy while familiarizing yourself with the interface/programming. If your approach is methodical—and you follow the flowchart, you shouldn’t encounter too many surprise results. Sound engine Polyphonic modulation sources Global control sources The user interface of Sculpture is broken down into three main areas. The large silver section at the top contains the sound engine.
Global Parameters These are found across the top of the Sculpture interface, unless otherwise specified. Transpose Transpose is used for coarse tuning of the entire instrument. Given the ability of component modeling to radically alter pitch with certain settings, coarse tuning is limited to octave increments. Tune Tune is used for fine tuning of the entire instrument (range: ±50 cents). A cent is 1/100th of a semitone.
If Sculpture’s string is still vibrating for a specific note, retriggering that same note will interact with the ongoing vibration, or current state of the string. Important: A true retrigger of the vibrating string will only happen if both Attack sliders of the amplitude envelope are set to zero. If either slider is set to any other value, a new voice will be allocated with each retriggered note.
String and Object Parameters The string and object parameters discussed in this section apply on a per-voice basis. You will note a number of parameter names followed by (morphable). This indicates that the parameters can be morphed between up to five snapshots, called morph points. More details on morphing can be found in “Morphing” on page 500. Hide, Keyscale, and Release Buttons This section briefly covers some common parameters that you’ll encounter when manipulating the string.
The Material Pad The following two string material parameters determine the general timbre, and are controlled by the ball (which correlates to the X and Y co-ordinates) within the Material Pad. The crosshair is a handle for the Key Scale and Release Scale diamonds in cases where these are hidden by the ball. It also allows you to independently change the keyscaling for one of the two axes (X/Y positions—Inner Loss or Stiffness).
Inner Loss Scale Release Values above 1.0 cause the inner losses to increase when the key is released. This is quite unnatural, as this would mean that the string material would change after the note was released. In practice, however, the use of this parameter in combination with Media Loss Scale Release allows a natural simulation of strings that are dampened at note-off time. To adjust, first enable the Release button, then click-hold on the blue Release line, and drag up/down to the desired position.
The String Parameters Around the Material Pad The following section covers the string parameters around the Material Pad. Resolution (Harmonics) This parameter determines the maximum number of harmonics contained in (and spatial resolution of ) the sound at C3. This is roughly proportional to the required CPU power, so the more harmonically-rich/the higher the Resolution setting of the sound, the more processing muscle will be required.
Media Loss Release The blue slider (in the outer ring of the Material Pad) controls the Media Loss Release time. To activate it, you must first enable the Release button, to the bottom right of the Material Pad. Values above 1.0 cause media losses to increase when the key is released. This parameter can be used to simulate a string that is dropped into a bucket of water after initially vibrating in air, for example.
Excite and Disturb Object Parameters The following parameters are used to excite, disturb, or dampen the string. Important: At least one object must be used, as the string itself does not make any sound! As you’ll discover shortly, there are a number of different string excite models such as blow, pluck, bow, and so on. Needless to say, these quite radically alter the general timbre of the string’s attack phase, resulting in bowed or plucked flute and bell sounds, or guitars with a flute-like blown sound.
The repositioning of objects changes the timbre of the string. If emulating, say a guitar, changing an object position could be viewed as similar to picking or bowing a string at various spots along a fretboard. About Objects and Velocity Sensitivity It is important to note that: Â Object 1 is velocity sensitive. Â Object 2 is only velocity sensitive when a type that actively excites the string is selected. When damping objects are used, Object 2 is not velocity sensitive.
Excite Types (Objects 1 and 2) The following table lists all excite types available for Objects 1 and 2, and information on the controls available for each.
Disturb and Damp Types (Objects 2 and 3) The following table lists all disturb/damp types available for Objects 2 and 3. 470 Name Description Disturb A disturb object that is placed at a The hardness of fixed distance from the string’s the object resting position Strength controls Timbre controls The distance from the resting position.
Gate Determines when the object is active—in other words, when it disturbs or excites the string. Settings are: Â KeyOn: Between note on and note off. Â Always: Between note on and the end of the release phase. Â KeyOff: Triggered at note off, and remaining active until the voice is released. Note: If using an object type such as Gravity Strike, the note may retrigger when you release the key. To avoid this artifact, set Gate mode to Always.
To adjust, simply click-hold and drag the corresponding numerical slider handle (the 1, 2, or 3 arrows) for each object. Adjustment of these object pickup positions will disturb/excite a given portion of the string. Object 1 can be an exciter. Object 3 can be a damper. You’ll note that Object 2 has two arrows. This indicates that this object can be used as either an exciter or damper.
 The vertical orange lines represent the positions of disturb/excite objects 1, 2, and 3. The thickness and brightness of these lines increases as the Strength level of each object is raised. Note: View these pickups as being like the electromagnetic pickups found on an electric guitar. Obviously, changing their positions will alter the tone of your axe, and they’ll do the same in Sculpture.
Pickup Spread Spreads the two pickups across the stereo or surround base. In other words, the pickup position, combined with this parameter, will be spread further towards/from the left/ right stereo/surround channels. Simply click-drag vertically on the Pickup button to adjust. Two dots in the ring that surrounds the Spread parameters indicate the values.
Decay Defines the decay time. The decay time is the amount of time that it takes for the signal to fall to the sustain level, following the initial strike/attack time. Sustain This parameter sets the sustain level. The sustain level is held until the key is released. Release This parameter determines the length of time that it takes for the signal to fall from the sustain level to a level of zero.
Input Scale (Morphable) This is a bipolar parameter. Negative values attenuate, and positive values amplify, the input signal prior to processing by the Waveshaper. When set positively, this results in a richer harmonic spectrum. The level increase introduced by the parameter is automatically compensated for by the Waveshaper. Given its impact on the harmonic spectrum, Input Scale should be viewed/used as a timbral control, rather than a level control.
Filter Type Buttons The five buttons at the bottom of the filter section determine the filter mode. Choices are: Â Hipass: Allows frequencies above the cutoff frequency to pass. As frequencies below the cutoff frequency are suppressed, it’s also known as a low cut Filter. The slope of the filter is 12 dB/octave in Highpass mode. Â Lowpass: Allows frequencies which fall below the cutoff frequency to pass. As frequencies above the cutoff frequency are suppressed, it’s also known as a high cut filter.
Velo Sens Determines the velocity sensitivity of the cutoff frequency. The harder you strike the keyboard you’re playing, the higher the cutoff frequency (and generally, the brightness of the sound) becomes. Â A value of 0.0 disables velocity sensitivity. Â A value of 1.0 results in maximum velocity sensitivity. Post Processing The post processing tools covered in this section impact on the summed signal of all voices, rather than on a per-voice basis.
In surround instances, the Xfeed control still controls the cross feedback between the delay lines, but offers additional cross feed modes. You can access these in Sculpture’s Extended Parameters section: LoCut Determines the cutoff frequency of the highpass filter at the delay line output/ feedback loop. HiCut Determines the cutoff frequency of the lowpass filter at the delay line output/feedback loop.
The Groove Pad (Stereo) When used in a stereo instance of Sculpture, the Spread and Groove parameters are combined in the two-dimensional Groove Pad. Drag the diamond in the center of the crosshair to adjust. You can independently adjust the Spread and Groove parameter values by directly dragging the lines that intersect the diamond.
The Spread parameter is accessible separately as a numerical edit field at the top-left of the pad. Click-drag, or double-click and type in, to alter the value. Clearing and Copying Delay Parameters You can Control-click on the Delay Time Pad to access a shortcut menu which offers Clear, Copy, and Paste commands for delay settings. These can be used to copy and paste delay settings between multiple Sculpture instances, or between consecutively loaded settings.
Basic EQ Controls The Basic EQ features different parameters than other EQ models. Please see the following section for details other model parameters. Â Â Â Â Low: Gain of a low shelving filter. Mid: Gain of a peak filter (sweepable—see below). High: Gain of a high shelving filter. Mid Frequency slider: Allows you to sweep the center frequency of the mid band between 100 Hz and 10 kHz.
Other EQ Models For all other Body EQ models you have the following parameters: Formant—Intensity Scales the intensity of the model’s formants. In other words, any formants (harmonics) in the model will become louder, or will be inverted, dependent on how this parameter is used. Â A value of 0.0 results in a flat response. Â A value of 1.0 results in strong formants. Â Negative values invert the formants. Formant—Shift This parameter shifts the formants logarithmically. A value of –0.
Please note that heavy use of Fine Structure may be quite CPU intensive. You should also note that the use of Fine Structure may not actually result in too much of a difference to your sound. This is dependent on the Waveshaper and Body EQ modes, plus other string parameter settings.
Level Limiter  Level: Controls the overall output level for the instrument.  Level Limiter mode: Clicking on the desired button activates or deactivates the integrated limiter. Options are:  Off: Disables the limiter.  Mono: A monophonic limiter on the summed signal of all voices.  Poly: A polyphonic limiter, that processes each voice independently.  Both: A combination of both limiter types.
 Two control envelopes can either be used as standard envelopes or as MIDI controlled modulators—with the ability to record, polyphonically play back (on a per-voice basis), and modify incoming MIDI controller movements. All modulation assignments take place within the generators. First order modulations are where the modulation generator modulates core synthesis parameters. Second order modulations are where one modulation generator modulates the parameters of the other modulation generator.
 The sawtooth is well suited for helicopter and space gun sounds. Intense modulations of incoming frequencies leads to bubbling and boiling, underwater sounds. Intense sawtooth modulations of lowpass filters create rhythmic effects.  The rectangular waves make the LFO periodically switch between two values (as an example—a positive value and zero—Unipolar). The Bipolar Rectangular wave switches between a positive and a negative value set to the same amount above or below zero.
 Curve values below 0.0: The slope at the zero crossing is reduced, resulting in shorter soft pulses to +1 and –1. Note: The waveform displayed between the Curve knob and the Waveform menu shows the results of these two parameter settings.
Target and Via Modulations Two target modulation destinations can be assigned per LFO, with an optional, additional via modulation. To activate, click on either the 1 or 2 buttons (which will highlight the Target and “via“ menus), then choose the desired target from the Target menu. The “via” menus determine the source that controls the modulation scaling for each LFO.
Waveform Menu Allows you to choose the waveform used for vibrato, for example sine, triangle, sawtooth, and so on. There are two special rectangular waves: Rect01 and Rect1—the former switching between values of 0.0 and 1.0 (unipolar), and the latter between values of –1.0 and +1.0 (bipolar, like the other waveforms). See “LFO 1 and 2” on page 486. Curve Allows you to define a freely-variable number of waveform variations, resulting in subtle/drastic changes to your modulation waveforms.
Random Variations Many sounds can benefit from the use of random modulations to parameters. These can emulate small variations that occur when particular instruments are played. Jitter 1 and Jitter 2 The two jitter generators are special LFOs, designed to produce continuous, random variations—such as those of smooth bow position changes. The jitter generators are equivalent to general purpose LFOs set to a noise waveform. To activate the routing of the jitter generators, click on the 1 or 2 buttons.
To activate the routing of the Note On Random modulators, click on the 1 or 2 buttons. Â Target: Determines the modulation destination—what parameter will be randomly modulated when a note is played. Â Amount: Sets the modulation amount—the strength of the modulation. Velocity Modulations The excite objects and the filter have dedicated velocity sensitivity controls. Many other modulation routings also allow you to select velocity as a via input source.
Controller A and B These parameters allow you to define two discrete modulation targets, and the strength of modulation for both Controller A and B. To use, simply click on the 1 or 2 buttons, select the desired target (and Target mode), and adjust the Intensity slider. Each target features a two-state button:  Continue: Continuous modulation  Note On: Modulation value is only updated when a note on message is received.
The envelopes can act as polyphonic modulation recorders and playback units. Each voice is handled independently, with a separate envelope being triggered as each note is played. To select Envelope 1 or 2, click on the Envelope 1 or Envelope 2 button. Modulation Routing As in the LFO section, each envelope offers two modulation target selectors with amount and via amount controls, and a separate via modulation option. The following target/amount/via settings are available for all run modes.
 The maximum time/length of the envelope is 48 bars/40 seconds.  The lines on the background grid are placed 100 milliseconds apart.  The background lines are placed 1000 ms apart for very long displayed envelope times. In sync mode, this is displayed as 1 quarter.  The envelope is zoomed automatically after releasing the mouse button. This allows the display of the entire envelope at the highest possible resolution for the graphic envelope display.
You cannot move a node beyond the position of the preceding node. You can, however, move nodes beyond the position of the following node—even beyond the right-hand side of the envelope display—effectively lengthening both the envelope segment and the overall envelope. When you release the mouse button, the envelope display will automatically zoom to show the entire envelope. To adjust the level of each node, click on the desired handle, and drag it up or down.
R(ecord) Button Enables the record functionality for the envelope. This button works in a similar fashion to the record arm buttons in Logic Pro. To stop recording, simply click on the R button a second time, or use the trigger mode functionality described below. Record Trigger Mode The menu to the right of the R button is used to choose different record trigger modes to start recording (when R(ecord) is active): Â NoteOn: Recording starts when a note is played.
Playing Back a Recorded Envelope Polyphonic playback of the recorded envelope occurs when you play a key. The Mode parameter must be set to Env and the R(ecord) parameter must be set to off. You can also activate both the Env and Ctrl buttons of the Mode parameter, as this will allow you to use controllers assigned to Ctrl Env1 or Ctrl Env2 to manipulate the envelope in real time, alongside playback of the recorded envelope.
When in any of the loop modes, the loop always cycles between user-defined envelope handles that indicate the loop start point (L icon), and the sustain point (S icon). These handles can be dragged to the desired position. Â When set to Finish, the envelope runs in one shot mode from its beginning to its end—even if the note is released before the envelope has come to its end. The other loop parameters are disabled.
VariMod—Source and Amount VariMod is only available for recorded envelopes. It allows you to select a modulation source, and amount, to control the strength of an envelopes variation. Variation in the envelopes means the deviation of a recorded envelope path from straight interconnecting lines between the points.
The morph section consists of two parts: Â The Morph Pad, featuring five morph points (center and four corners), plus options for randomizing (via the randomize parameters) and coping or pasting morph points and Morph Pad states via a shortcut menu. Â The morph envelope, that can be edited either by segment (with the mouse), or recorded via MIDI controller movements. With a vector stick (Morph X/Y controllers) or mouse movements of the morph cursor (the ball), for example, on the Morph Pad.
Randomizing Morph Points The randomize feature allows you to create random variations of selected morph points. When combined with the copy/paste functionality that’s also available, randomizing lends itself to using the Morph Pad as a kind of sound cell culture device. Use of the Morph Pad can yield an interesting composite sound. You can copy this sound to a corner of the Morph Pad (or several corners) and randomize it by a definable amount.
5 Click-hold on the morph cursor (the ball), and drag it to each of the corners in the Morph Pad. Do this along the edges, as well as through the center of the Morph Pad, and take a mental note of how this impacts on the morph. 6 Don’t forget to strike a few notes on your MIDI keyboard while doing so. Note: The morph ball is only visible when the Record Trigger button is active.
Other Commands The remaining commands in the Morph Pad shortcut menu are to do with grouping of random parameters. Put another way, the following commands allow you to determine which type of parameters you would like to randomize (via the Rnd button and Int(ensity)) slider. Â All (Random Group): This is your ticket to wacky sounds as all parameters in the following three groups are randomized. This can lead to some interesting results, but can be uncontrolled.
 As you move your mouse cursor along the line, or hover over the nodes directly, the current envelope segment is highlighted.  You can create your own envelopes manually, by manipulating the nodes and lines, or you may record an envelope, as discussed in “Recording Morph Envelopes” on page 505.  To adjust the time between nodes, click on the desired handle, and drag it left or right. As you do so, the overall length of the morph envelope will change—with all following nodes—being moved.
 Note + Sustain Pedal: Recording starts when the sustain pedal is depressed while a note is held. Recording is stopped by clicking the R(ecord) Enable button (or trigger) a second time. Once all keys are released, and all voices have completed their decay phase, the recording ends. You can stop recording early by releasing all keys, and then pressing a single key. Following the recording of a controller movement, R(ecord) Enable is automatically set to off and Mode is set to Env only.
Time Scale This parameter scales the duration of the entire envelope between 10% and 1000%. Sustain Mode Menu Allows you to select the behavior of the morph envelope while a note is held. Choices are: Sustain mode, Finish mode, one of three loop modes (forward, backward, alternate), or Scan via CtrlB mode. When in any of the loop modes, the loop always cycles between the loop and sustain envelope handles (the nodes indicated by the small L and S icons).
∏ Tip: It is also possible to manually drag the red time position marker with the mouse. Note: If one of the three loop modes is selected, and the loop point is positioned before the sustain point, the loop will be active until the key is released. Following key release, the envelope will then continue beyond the sustain point, as per usual.
Morph Position Display The red line in the morph envelope’s timeline shows the current time position, during a morph. Sculpture’s Morph Pad displays a moving dot that indicates the current morph position. The dot and line indicate the current morph position. Note: The current morph position is shown as long as only one note is played. MIDI Controller Assignments This section allows you to define the MIDI controllers you wish to use for vibrato depth control or morph pad movements, for example.
CtrlEnv 1 and CtrlEnv 2 Menus Set the controller assignments for the two control envelopes—used as a modulation signal or an offset—in cases where the control envelope is set to Ctrl only or Ctrl+Env modes. It also is used to define the source for recording controller movements. Morph X and Morph Y Menus Determine the controller assignments for the X and Y co-ordinates of the Morph Pad.
Programming: Quick Start Guide This section of the manual contains a collection of programming guidelines, tips, tricks, and information to assist you in creating particular types of sounds. A more involved look at programming can be found in “Programming: In Depth” on page 527. Approaches to Programming Given the flexibility of Sculpture’s synthesis core, you can take a number of different approaches to sound design.
The Core Engine The signal flow was discussed in “The Synthesis Core of Sculpture” on page 456. To recap, and explain in a more hands on way, please follow these steps. This section is intentionally simplified, but please work through it. Knowing the mechanics of Sculpture is essential to your success: The String The string is the central synthesis element.
 You probably noticed that moving the Media Loss, Tension Mod, and Resolution sliders also had an impact on the green and blue Keyscale sliders inside and outside the ring. Grab, and drag each of these keyscale slider arrowheads to different positions—one by one—and play a few notes either side of middle C as you’re doing so. Note the changes that happen up or down the keyboard range.
 Change parameters for this object as desired, and note the interaction of the two objects with each other and the string.  Do the same for Object 3. The Pickups The vibration of the string is captured by two movable pickups. The Pickup section also houses three sliders—one for each object.  Reload the “plain vanilla” setting.  Click-hold on the Object 1 pickup handle—the down arrowhead with the number “1” on it—and drag it left or right while striking a key.
In effect, each parameter that you introduce or make changes to, will affect the modelled string. This will, in turn, affect the interaction of each parameter with the modelled string. As such, parameter settings that you have already made for, say Object 1, may need to be adjusted when Object 2 is activated. Generally, such adjustments won’t need to be radical, and may only involve a small tweak to the Strength parameters, or perhaps the pickup positions of each object, for example.
All other parameters on the lower portions of the Sculpture interface (Modulation, Morph, Envelope, and Controller Assignments) are not part of the core synthesis engine, although they can obviously impact upon it. Some tips and uses of these parameters are discussed below. Creating Basic Sounds This section covers the creation of basic types of sounds, such as organs, basses, guitars, and so on.
Is it polyphonic or monophonic? This is a pretty significant factor, that ties into the next question. Apart from the obvious things such as the inability to play chords on a flute, a modelled string will interact with any currently active string. This, of course, can’t happen in a flute. It’s strictly a one-note instrument.
Basses Creating bass sounds with Sculpture is pretty straightforward. Â Load up your “plain vanilla” setting. Â Set the Transpose parameter to 1 Oct., and play a few notes around C2. You’ll note that the general color of an acoustic bass is already there. Â You can certainly adjust the ball on the Material Pad towards the Nylon entry, but before doing so, change Object 1’s type to Pick. Â Have a play on the keyboard, and adjust the ball position while doing so.
 Activate the Body EQ by clicking the Body EQ button. Ensure that the Lo Mid Hi model is selected.  Adjust the Low level to 0.55, the Mid to 0.32, and the Hi to 0.20.  At this point, you will have a working bell sound—but—you’ll probably find that there is a tuning issue below C3, in particular. This programming approach was taken because the harmonics of the sound are most noticeable after all other parameters have been set.
 Save setting as… with a new name. Please explore this patch much further. There are a great number of directions that it could be taken in—as a muted trumpet, french horns and even sitars or flutes. The Waveshaper has a significant impact on this sound, and this is the first place you should look to radically alter it. Use the Delay to create space and the Body EQ to cut the lows and boost the Mids and Hi’s.
 A number of approaches can be taken to add interest to the sustained sound. These include; using the vibrato modulator (assigned to aftertouch, perhaps), or perhaps recording or drawing in an envelope, and controlling the Waveshape Inner Scale via Velocity and/or String Media Loss. You could even use the Loop Alternate Sustain Mode. Feel free to experiment!  Save setting as… with a new name.
Obviously, this is but one guitar sound. You can make use of the Object Strength, Variation, and Timbre parameters, not to mention repositioning the Material Pad ball to create a completely different tone to your guitar. For quick and easy mandolins, make use of the Delay (or Vibrato) to emulate the double-strike picking that is associated with the instrument. Organ Organ sounds are amongst the easiest and quickest sounds to emulate in Sculpture as they have no release phase.
At this point, you should have a basic organ tone. Save setting as… with a new name. You can use this as the basis for your next organ patch. You’ll probably notice some intermodulations that are introduced when you’re playing chords. Apart from the pitch differences between notes in the chord, this is a result of the interactions between each voice being produced by Sculpture.
 Set the Object 1 Velosens slider to match your playing style and that of the music, as you’re playing the keyboard. Adjust later, if desired.  Grab the Tension Mod slider, and move it slightly upwards, so that the arrowhead covers the “D.” This emulates the momentary detuning effect of the bow stretching the string.  Move Pickup A to a position around 0.90.  Move Object 1’s pickup position to a value around 0.48.  Activate the Body EQ, and select the Violin 1 model.
     Play a “C” chord (middle “C”), and you’ll hear a pad sound. Move Pickup A to a position around 0.75, and the pad will become a little sweeter. Move Object 1’s position to a value of 0.84. Move Object 2’s position to a value of 0.34. As a final step, click on the Points icon that features five dots in the Morph Pad section.  Set the Int slider in the Morph Pad Randomize section to a value of say 25%.  Click once on the Morph Rnd button.  Save setting as… with a new name—say “vanilla_pad.
 Now change the Morph Mode to Env only, and you should see your Morph circle.  Play the keyboard. There’s your morphed pad!  Feel free to adjust the morph envelope parameters. Remember when you were asked to use the Morph Points, Intensity, and Rnd parameters while setting up the “vanilla_pad”? This was to ensure that there were several morph points already available for your morphing pleasure.
Programming: In Depth This tutorial explains how you can program sounds with Sculpture from scratch. Based on Sculpture’s string model, you’ll learn how to use the individual sound shaping parameters in order to recreate the physical properties of an instrument in detail. Note: You will find these tutorial settings in the Factory > Tutorial Settings sub-folder of the Settings menu (in the header of the Sculpture plug-in window).
The number of frets differs from bass to bass and depends on the scale length. Don’t worry about pitches higher than a single ledger line C; the actual functional range of this instrument is primarily in its two lower octaves—between E 0 and E 2. Also worth mentioning is the fretless electric bass. Like all instruments of this type, it is freely tunable and possesses a distinctive, individual sound. Over the course of this tutorial you will discover how to program this type of instrument sound in Sculpture.
The vibration of the strings is, of course, naturally hampered by several physical factors: the radius of motion of the string (antenode) is impeded by the left bridge or by the first fret that’s pressed down upon (and the frets in between). This can lead to the development of overtones which can take the form of anything from a slight humming or buzzing to a strong scraping or scratching sound.
Play some notes in the lower range. you’ll note that the sound is very muffled, hollow, and distorted. Before you adjust further parameters in Object 1, you need to set the position of the pickup. This is accomplished in Sculpture’s pickup display located to the left of the Material Pad. You’ll find three trapezoidally shaped sliders, representing Objects 1 to 3. Both of the transparent bell-shaped curves help you to visualize the position and width of Pickup A and Pickup B.
Although you can already recognize the sound of an electric bass, it doesn’t sound wiry enough yet. Now it’s time to focus on the bass strings themselves. In order to recreate the material properties of a set of round wound strings: 1 Move the ball in the Material Pad up and down at the left edge. Pay attention to how the overtones react. Move the ball to the lower left hand corner. The sound should vaguely remind you of the sound of a low piano string.
The vibration of a bass string does not occur in a vacuum. The antenode of the string frequently encounters the natural, physical limitations of the instrument. This is heard as the typical buzzing and rattling that occurs when the strings touch the frets. To simulate these disturbing elements with Object 2: 1 Activate Object 2, and choose the Bouncing type menu item. The sound should now vaguely remind you of a mandolin tremolo. This is way too strong an effect for this kind of sound.
Note: The most relevant performance range for basses is found exclusively below C3. For this reason, you should make use of the green sliders to set the actual timbre of the sound. The primary sliders found around the ring determine the timbre of the sound above C3. For the moment, ignore the blue sliders (which control high key-scaling) and simply set them to the same positions as the main sliders. Once activated, the key-scaling function is used to adjust the timbre of the sound, independent of pitch.
Use the Inner Loss parameter to scale the overtone content, dependent on pitch: 1 Move the Material Pad ball above the words Inner Loss. Try to move the ball solely in a vertical direction, in order to maintain a constant Stiffness value. 2 Grab the green line next to the ball, and pull it towards the bottom until the small green diamond is located directly above the word Steel.
You’ve completed this section, and created a basic bass that’s articulated with your fingers. Save this as E-Bass Fingered Basic. In the following sections, you’ll be using this basic bass as a foundation for the construction of further bass sounds. Modifying the Frequency Spectrum of Your Basic Bass The scope for sound design, by altering the frequency spectrum of electromagnetic instruments, is far more flexible than that offered by acoustic instruments.
7 To finish off, set the Level dial (to the right of the amplitude envelope) to a value of – 3 dB. The sound is now as loud as possible, without the low notes distorting. 8 Save this sound setting, as you’ll need it for further modifications later. Please name it E-Bass Fingered Basic EQ1. Pick Bass The basic bass is played with the fingers. In the following example you will simulate playing the strings with a pick, using the Pick object type.
Damping Playing with a pick is often combined with a damping technique that employs the ball of the thumb. The right hand, which also holds the pick, should physically lay on top of the strings at the bridge. This technique results in the sound having less overtone content but become more percussive and punchy at the same time. You can variably control the timbre of the sound through the angle and pressure of your hand while playing.
To simulate fingers lightly touching the strings: 1 Object 3 is used as a damper. Select the Damp type. 2 Adjust Object 3’s Timbre parameter to its maximum value of 1.00. 3 Variation must be set to its initial value of 0.00. Simply click on the slider while holding down the Option key to do so. 4 Move Object 3 to the exact middle (0.50) of the Pickup display. Play the keyboard, and you’ll hear the first overtone as a harmonic.
5 To remove the smacking in the attack phase, use the graphical display to choose a value of 0.48 for the Body EQ Mid frequency, then use the dial to increase this value to 0.51. Option-click on the Body EQ High parameter to set it to a value of 0.00. 6 Save this setting as Easy Listening Pick Bass. Slap Bass You’re actually dealing with two different articulations here. The low notes originate when the thumb literally slaps the strings on the upper part of the fingerboard.
8 Move Object 1 to position 0.90 in the Pickup display. This position corresponds to a playing position above or on the fingerboard. Note: Given its universal concept, Sculpture will not react exactly like a bass, where one would tend to play in the middle of the string on the upper part of the fingerboard. Try moving Object 1 to this position and see how it sounds. You’ll find that the sound is little too smooth. Setting the parameters for Object 1: 1 Set Timbre to a value of 0.
Fretless Bass With the exception of shared playing techniques, the fretless bass differs from a normal bass through its buzzing, singing sound. As the frets on the fingerboard of a standard bass function as a collection of mini-bridges and allow the string to vibrate in an unobstructed fashion, the direct collision of the string’s antenode with the fingerboard on a fretless bass is responsible for its typical sound.
8 Click on the button marked 1 (next to the Rate slider, to the upper right) to activate the first modulation target. 9 Choose Object2 Strength as the Target parameter. 10 Select the KeyScale entry in the via column. 11 Move the lower slider labeled Amt (amount) to the right while you are playing. You’ll quickly realize that the singing buzzing fades out in the lower range, while gradually being retained as you move towards C3. Set the slider to a value of 0.15.
You can hear how the stereo breadth of the fretless sound has increased. Pickup A is sent out on the right channel, while Pickup B occupies the left channel. Note: Although only modern basses offer such stereophonic features, it’s still fun to process conventional sounds (such as those created in the previous examples) with this effect.
Reverb and Reflections As a rule, basses are mixed without effects (dry) and you probably haven’t missed any reverb or delay effects in the examples, so far. Having said this, a little bit of reverb can be quite appealing on a fretless bass, when it’s used as a solo instrument. Use Sculpture’s Delay section to emulate this. In order to create an unobtrusive atmospheric space, proceed as follows: 1 Load the Fretless Chorus Dry setting. 2 Turn on the Delay section by clicking on the Delay button.
Creating a drowned in delay effect: 1 Reload the Fretless Chorus Dry setting. 2 Switch the Delay section on. 3 Move the Input Balance slider all the way to the right (1.00). 4 Set the Delay Time value to 1/4t (quarter note triplet). 5 Set the Feedback dial to a value of 0.20. 6 Adjust the Xfeed knob to a value of 0.30. 7 Set the LoCut to 200 Hz and the HiCut to 1600 Hz. 8 Now adjust the overall level of the effect; try setting the Wet Level dial to a value of 45%.
Within the framework of these short experiments, it’s of course impossible to comprehensively cover all of Sculpture’s possibilities. Please experiment with the suggested settings and closely observe the results of the changes you make. In this way you can learn a lot about the instrument, and hopefully be inspired to create new sounds and variations. Note: You will find the settings for these tutorials in the Tutorial Settings folder in the settings menu (in the head of the Sculpture plug-in window).
Proceed as follows to record an envelope: 1 Move the Object 1 slider all the way to the left. Starting from this position (where it only generates an overtone-rich scratch), start animating it by using the envelope. 2 Locate the Envelope section in the lower right corner of the Sculpture window. Choose the first of the two envelopes by clicking on the envelope 1 button, if necessary.
You can now see the curve you recorded in the graphic display. You’ll note that the curve arches exclusively above the zero axis—this is no surprise as the modulation wheel only sends unipolar values, which means between zero and up to a positive maximum value. As the Object 1 slider has already been moved all the way to the left end of the string, it can only be shifted all the way to the right by the envelope when the maximum modulation intensity is reached.
Increasing Stereo Breadth and Chorus To give this very dry sounding pad a little more stereo breadth and chorus effect, use the trick discussed while creating bass sounds. To refresh your memory, modulate the Pickup positions, and assign them to the left and right channels. Here’s a quick description of the process: 1 Click-hold on the Pickup semicircle in the Spread control element, and move the mouse upwards to separate the stereo pan positions of the Pickups.
To give the pad a little depth, activate the Delay. Set the Delay Time to 1/4 and adjust the Xfeed knob to 30%. The pad now has a pleasant and unobtrusive ambience; you can leave the other Delay parameters at their original values (preset 0003 eqfx pad). Finally, to optimize the sound so that it is a little more animated. The end result you should be aiming for is subtle, which makes the jitter modulators the perfect tool for the job. The jitter modulators are basically LFOs that use a random waveform.
You now have a satisfactory pad sound, which you should leave alone at this point, even though a few Sculpture features such as; the Filter and the Waveshaper lie idle, not to mention the two additional Objects—but sometimes it’s smart to quit while you’re ahead. The last function, morphing, has been saved for the end. Follow this section to bend and twist your pad sound (a little or a lot). Morphing In the middle of the lower part of Sculpture’s window you can see the Morph Pad.
Surrounding the Morph Pad, you’ll find a randomizing function which randomly varies sounds to a chosen intensity level (or amount of randomization). This is especially useful for subtle changes to natural sounds, but it can also provide for rewarding variations to synthesized sounds as well. To use the randomizing function: 1 On the left side of the Morph Pad, choose the number of corners that are to be varied by selecting one of the cubes.
29 Ultrabeat 29 Ultrabeat is a rhythm synthesizer that incorporates a step sequencer. Ultrabeat’s synthesis engine is optimized for creating electronic and acoustic drum and percussion sounds. It includes an impressive diversity of synthesis engines: phase oscillators, sample playback, FM (frequency modulation), and physical modeling.
The Structure of Ultrabeat Most software synthesizers offer one synthesizer per plug-in instance. Ultrabeat, however, places 25 independent synthesizers at your disposal. These synthesizers— called drum voices in Ultrabeat—are optimized for the generation of drum and percussion sounds. The distribution of drum voices across the MIDI keyboard is simple and easily explained: the first (starting from the bottom) 24 MIDI keys are each assigned a single drum voice.
Overview of Ultrabeat Ultrabeat’s user interface is divided into three functional sections. Synthesizer section Assignment section Step sequencer  Assignment section: Displays all drum sounds of a drum kit, allowing you to select, rename, and organize them. It also includes a small mixer, which you can use to adjust each sound’s volume and pan position.
Loading and Saving Sounds You can use the same methods to save and load settings in Ultrabeat as in all other Logic Pro instruments. For more information, see the Logic Pro 8 User Manual. An Ultrabeat setting contains: Â The drum kit, which consists of 25 sounds, inclusive of assignment and mixer settings. Â The complete settings of all parameters for all 25 sounds. Â The sequencer settings and all 24 patterns, including the step automation, trigger, velocity, and gate rows for all 25 sounds.
The Assignment Section The Assignment section displays all drum sounds of a drum kit, and also includes a small mixer. It allows you to:  Select, organize, and name drum sounds  Import drum sounds from other Ultrabeat settings or EXS instruments  Mix drum sounds Selecting Sounds The 25 sounds of an Ultrabeat drum kit are mapped to the onscreen keyboard found on the left hand side of the Ultrabeat window.
Naming and Organizing Sounds Double-clicking on the name of a drum sounds opens its text entry field, allowing you to rename it. Press Return or click anywhere outside the text entry field to complete the naming operation. Swapping and copying drum sounds within an Ultrabeat kit can be achieved via a drag and drop operation or via a shortcut menu. To swap or copy drum sounds using drag and drop: 1 Click-hold the drum sound in the Assignment section (not on a button or menu).
 Copy (Voice & Seq): This command copies the selected sound—inclusive of Mixer settings and sequences—to the Clipboard.  Paste Voice: This command replaces the selected sound with the sound from the Clipboard, without changing its sequences.  Paste Sequence > (submenu): The Paste Sequence submenu allows you to replace all, or individual sequences, of the target drum sound. Paste Sequence has no impact on the sound parameters of the drum sound.
A list of all the sounds found in the selected setting or EXS instrument will be shown next to the Mixer section. Note: If you import EXS instruments that include more than 25 sounds, you will be able to page through sets of 25 sounds using the up and down arrows to the left and right of the EXS instrument name at the top of the import list. There are two methods to transfer sounds from the import list to the Assignment section.
 Paste Sequence > (submenu): Use to replace all, or individual sequences, of the target drum sound. Paste Sequence has no impact on the target drum sound. Pasting a single sequence replaces the currently active sequence (as set in the Pattern menu) of the target drum sound. This allows you to import drum sound sequences into any of the 24 possible pattern locations. Note that these operations only affect the selected drum sound, all sequence and sound data of the other 24 drum sounds remain untouched.
Mute You can mute individual sounds in a drum kit by clicking the Mute button (M) to the right of the name. Solo You can listen to sounds in isolation by clicking the Solo button (S), found beside the Mute button. Pan The rotary knob to the right of the Mute and Solo buttons controls the placement of the signal in the stereo field (Panorama). Individual Outputs Ultrabeat features eight separate stereo and mono outputs, and can be inserted as a multi output instrument.
The Synthesizer Section The Synthesizer section is the heart and soul of Ultrabeat. As noted above, each drum sound has its own Synthesizer section. Don’t be intimidated by how many parameters the Ultrabeat synthesizer squeezes into one plug-in window; in fact, its signal flow is quite easy to understand. The Signal Flow Ultrabeat’s synthesis engine is based on classic subtractive synthesis principles.
The filter receives its signal from the following sound sources: Oscillator 1, Oscillator 2, the noise generator and the ring modulator. Their output sections are displayed by four objects that sit adjacent to the filter (three round objects and the smaller, rectangular ring modulator to the right of the filter). One level down, you’ll find the control elements for these sound sources.
The pitch value is displayed to the left of the slider. You can change the displayed value by click-holding directly on the value field, and moving the mouse vertically. Pitch can also be modulated by the sources found in the Mod and Via menus. You’ll find a signal flow switch between each of the oscillators and the Filter section that controls routing (filter bypass button).
 Saturation: Increasing Saturation values clip the waveform, gradually molding its shape towards a rectangular waveform. This results in a corresponding increase in odd numbered overtones.  Asym (Asymmetry): Tilts the waveform towards a sawtooth wave, making the sound more edgy. Asym can be modulated by the sources found in the Mod and Via menus. This allows you to create dynamic sound changes at the oscillator level. For more information, see “Modulation” on page 581.
Once you select Side Chain mode for Oscillator 1, you need to select which channel you wish to use as your side chain input. You do this in Ultrabeat as you would with any plug-in with side chain input by choosing the desired channel in the Side Chain menu. Side Chain mode greatly expands your creative options when using Ultrabeat. You can use an audio input from Oscillator 1 along with the synthesis engine of Oscillator 2 to create a part live audio, part synthesized drum sound.
Sample In Sample mode, Oscillator 2 uses an audio file as sound source. Clicking the arrow in the upper left corner of the waveform display opens a menu, that allows you to load and unload samples, or to display the loaded sample in the Finder. The Reverse arrow changes the playback direction of the sample (forwards/backwards). The two Min/Max (Velocity) horizontal sliders determine the start point of the sample—dependent on the dynamics of the performance.
A selection of multi-layer drum and percussion samples that were specially created for Ultrabeat and its function set are included with Ultrabeat. You can also load your own samples in AIFF, WAV, CAF, or SDII stereo interleaved format. It should be noted, however, that the velocity layering function is not available for such samples. The Play button in the Load Sample window allows you to preview audio files (AIFF, WAV, SD2, CAF, UBS) before loading.
When saving a drum kit using the Settings menu, the location of the sample is saved with the setting. The Ultrabeat setting doesn’t actually save the audio files themselves—only a reference to their location. If you load a setting that contains a reference to a sample that has been moved or erased, Ultrabeat will present you with a dialog box that requests you to find it. To avoid this problem, it is highly recommended that you use a dedicated Ultrabeat sample folder.
To the right of the Material Pad you’ll find the Resolution parameter. In contrast to the other parameters of the Model oscillator, Resolution does not reproduce a pre-defined real-world property of the physical model, but affects the modeling process itself: higher values lead to an improved calculation resolution which results in more overtones. Lower values reduce the precision of the calculations, leading to fewer overtones and often to inharmonic spectra.
Note: The filter bypass button determines the signal flow. It doesn’t turn the ring modulator on or off. Use the Ring Mod field for this (see above). The Noise Generator The fourth synth engine is the noise generator. Noise contains—in a technical sense—all tonal frequencies; that’s why human hearing can’t recognize any tonality in a noise signal. Despite this (or as a direct result of it), noise is an indispensable ingredient when creating drum sounds.
The bandpass (BP) filter only allows a certain frequency range (a frequency band) centered around the cutoff frequency to pass. It can be used in the upper, as well as at the lower, end of the frequency spectrum to reduce the highs and lows of a sound. The Cut knob determines the cutoff frequency, defining the point in the frequency spectrum where reduction begins. Depending on the type of filter you select, you can make a sound darker (LP), thinner (HP) or more nasal (BP) by adjusting the Cut value.
The Filter Section The output signals of both oscillators, the ring modulator, and the noise generator are passed on to Ultrabeat’s central Filter section (if they haven’t bypassed it through use of the various filter bypass buttons). The Filter section offers a multimode filter and a distortion unit. Multimode filter Distortion unit The order that sounds are passed through the filter and distortion unit is determined by the arrow found at the “equator” of the Filter section.
The names of the individual filters illustrate their function: A lowpass (LP) filter allows frequencies lower than the cutoff frequency to pass. It removes (cuts) the highs of a sound, making it darker and less bright. A highpass (HP) filter allows frequencies higher than the cutoff frequency to pass. The lows of the sound are cut. A bandpass (BP) filter allows a frequency band centered around the cutoff frequency to pass.
The Distortion Unit Depending on the order determined by the arrow in the Filter section, the distortion unit is inserted either before or after the multimode filter. It provides either a bit crusher or distortion effect. The desired mode is activated by clicking on the Crush or Distort button. The active effect is indicated in red. If neither button is red the distortion unit is bypassed The distortion effect is modeled on an analog distortion unit, which distorts the sound by overdriving the level.
Two Band EQ Both equalizer bands have almost identical features. Their parameters are explained jointly, but you can, of course, adjust band 1 (the upper EQ in the Output section) and band 2 separately. Clicking on the Band 1 and Band 2 labels turns the individual band on or off. When active, the field is red. If neither EQ is activated, the signal passes through unaffected. The EQ type buttons switch between two different types of EQs: shelving and peak.
Note: Option-clicking the Hz parameter returns its value to a neutral position. This is 200 Hz for the first band and 2000 Hz for the second. The selection of these default frequencies was made in accordance with the different shelving characteristics of each frequency band. Band 1 is designed to filter low frequencies and band 2, high frequencies. The Q factor is regulated by click-dragging vertically on the Q parameter field.
Pan Modulation Pan Modulation varies the panorama position of a drum sound dependent on a Mod and Via source. The modulation set here is relative to the panorama position set in Ultrabeat’s mixer. The panorama position set in the mixer is represented here by a thin, red line. To the left and right of the line, small sliders (and corresponding menus) allow the adjustment of the Mod and Via modulation routings.
Note: The leveling stage for Voice Volume precedes the sliders in the mixer. This approach allows the starting volume of the individual drum sounds to be set independently of their relative levels in the drum kit mix. Trigger and Group Menus The manner in which Ultrabeat reacts to a succession of incoming notes is individually defined for each sound. These parameters are found in the Output section, below the Voice Volume knob.
Modulation Numerous sound parameters can be controlled dynamically (modulated) in Ultrabeat. Ultrabeat provides two powerful LFOs, four envelope generators, velocity, and four freely-definable MIDI controllers as modulation sources. The setting of modulation routings follows a universal principle that is explained in this chapter.
The Cut (Cutoff ) parameter has a mean (default) value of 0.50. It’s not being modulated yet as no modulation source has been selected in either the red Mod or blue Via menu (set to Off ). As soon as a modulation source is selected in the Mod menu (Env 1 in this example), the ring around the rotary knob is activated. Grabbing and moving this ring with the mouse allows you to set the value that this parameter will be increased to by the Mod source (0.70 in the example).
Back to the example; the frequency of the filter is set to the mean value of 0.50. When the Mod source Env 1 enters the equation, the Env 1 envelope generator drives the Cut value up from 0.50 to 0.70 (during the attack phase) and back down to 0.50 (during the decay phase). Note: You can view the exact values in the help tags that appear when you grab the individual handles of various parameters.
Setting the Modulation Routing Clicking on the Mod label opens the Mod menu. This is where you can choose one of the LFOs or envelope generators (Env) as a modulation source. The Off setting deactivates the Mod routing, and the Mod ring can no longer be adjusted. In this situation, no Via modulation can occur either (this is because Via no longer has a modulation target) and the Via slider disappears. Note: The Max setting produces a static modulation at maximum level.
MIDI Controllers A–D In the MIDI Controller Assignments area at the upper edge of the Ultrabeat window you can assign a standard MIDI controller to each of the four controller slots: Ctrl A, B, C, or D. Ctrl A, B, C, and D can be used as Via modulation sources within Ultrabeat. Use these assignments to set up your external MIDI controller hardware to operate with Ultrabeat. As examples: aftertouch or the modulation wheel of your MIDI keyboard.
The LFO section display shows the LFO waveform, the shape of which is governed by the Shape slider located underneath it. Dragging the slider from left to right causes the waveform to fluidly morph from a sine to a triangle, and then finally to a square wave (with variable pulse width), including all variations in-between. At the far right hand position of the Shape slider, the LFO produces random waveforms. The LFO speed (Rate) can be set independently (Free) or synchronized (Sync) to the Logic Pro tempo.
Env 1 to 4 Further modulation sources available to you in the Mod menu include four identically specified envelope generators. Envelope parameters are described in this section. Note: In addition to its potential use in the Mod menus of various sound parameters, Env 4 is permanently connected to the Voice Volume. In other words, Ultrabeat has a hard-wired volume envelope generator.
Envelope Parameters In order to edit the envelope parameters, first select an envelope by clicking on the desired 1 to 4 buttons. The parameters of the corresponding envelope can now be directly changed in the envelope display window. Attack Time Attack time defines the period of time the envelope needs to reach its maximum value. This is measured from the instant you press a key (note on). This period is called the attack phase.
Note: If the Sustain button is not activated, the envelope functions in one shot mode, and the note length (MIDI note off command) is disregarded. Zoom (to fit) When you select the Zoom button, the envelope is enlarged to fill the entire width of the display, making it easier to adjust junction points and curves. The graphic display is quickly redrawn after any change is made to the Attack or Decay values.
The Step Sequencer The integrated step sequencer allows all Ultrabeat sounds to be combined in sequences, based on patterns. Its design and use (step programming input) are based on analog predecessors. Unlike Ultrabeat’s analog predecessors, however, you can also program automated changes for nearly every synthesizer parameter! Dependent on your personal taste and favored musical style, you’ll want to control Ultrabeat from either the integrated step sequencer or from Logic Pro, when programming rhythms.
Step Sequencing With Ultrabeat Ultrabeat’s step sequencer contains 24 sequences—each consisting of up to 32 steps. The sequencer is divided into three sections. Global parameters Pattern parameters Pattern parameters Step Grid  Global parameters: These parameters globally control the pattern and sounds, independent of the individual steps and patterns.  Pattern parameters: Control the currently selected pattern.  Step Grid: Here, the actual sequencing takes place.
 Step mode: In Step mode, you can automate a sound’s parameter from one step to the next. Values are offset, instead of set—all of your original drum sound settings remain unaltered by any parameter changes performed in Step mode. Step automation changes only operate on parameters when the sequencer is running. These parameter changes occur individually per step. This means that if the sequencer is turned off, you’ll hear the original sound. For more information see “Step Mode” on page 597.
Resolution This parameter determines the resolution of the pattern. It defines the metric unit of a measure that is represented by the individual steps. As an example: The 1/8 setting means that each step of the grid represents an eighth note. Given a pattern length of 32 steps, the pattern would run for 4 measures (32 ÷ 8). The Resolution setting applies to the entire grid, and therefore, equally to all sounds.
Step Grid In the step grid, the pattern is displayed in numerous rows and steps. The rows always correspond to the sound that is currently selected in the assignment area. Choosing a different sound switches the sequencer display to show the rows that correspond to the newly-selected sound. The step grid area contains two rows—each with 32 fields. Trigger row Velocity/Gate row  Trigger row: Click a button to activate or deactivate the sound on that corresponding beat.
 Add Every Downbeat: Adds triggers on every downbeat until the sequence is filled. The exact determination of which steps are downbeats depends on the grid resolution. For example, if the resolution is set to 1/16, Add Every Downbeat would create triggers on every 4th step. Starting with the initial downbeat at step 1, this would create trigger events on step 5, step 9, step 13, and so on. This command does not erase existing trigger events, it only adds trigger events.
 Create & Replace Some: Same as Create & Replace Randomly above, however it creates less trigger events. How many events are created depends on the grid resolution.  Create & Replace Many: Same as Create & Replace Randomly above, however it creates more trigger events, filling the sequencer with events. Velocity/Gate Row In this row, you set the length (gate time) and the velocity of the notes entered in the trigger row. Both parameters are displayed as a single graphical bar.
Switching the Step Grid to Full View Clicking the Full View button in Ultrabeat’s lower right corner switches the synthesizer controls to a large grid filled with trigger buttons. The large grid displays the 32 trigger buttons for each of the 25 drum sounds simultaneously—independent of the currently selected sound. As usual the selected sound is visible in the step sequencer area so that you can set the velocity and gate time for each step as well as offsets in Step mode.
All parameters you automate will show up in the pop-up menu at the top of the Parameter offset row (see “The Parameter Offset Row” below). Offset menu Parameter offset row When you engage Step mode, Ultrabeat’s interface changes in the following ways: Â Yellow frames appear around all parameters in the Synthesizer section that are available for automation. Parameters not available for automation are still visible, but are disabled.
Entering Offsets By clicking in the parameter offset row you select the step into which you would like to enter a value. Every following parameter change made in the Synthesizer section will be recorded as an offset value for this step in the parameter offset row. To provide a better overview, the parameter offsets in the Synthesizer section are displayed by a yellow range of values. To enter an offset for a new parameter, click-drag on any yellowoutlined parameter in the Synthesizer section.
Note: Moving a control element in the Synthesizer section whose value has not yet been changed in step edit mode adds an additional entry to the offset menu. Parameter Offset Row Buttons The Parameter offset row has three buttons: Mute, Solo, and Reset The buttons functions are:  Mute: Mutes the offsets of the selected parameter  Solo: Solo the offsets of the selected parameter  Reset: Returns all the offset values of the selected parameter to zero (no offset).
To clear a pattern: 1 Select the desired pattern in the Pattern menu. 2 Control-click (or right-click) the Pattern menu, then choose Clear in the shortcut menu. Exporting Patterns as MIDI Regions Patterns programmed in Ultrabeat’s internal step sequencer can be exported as MIDI regions into the Arrange area of Logic Pro. To export an Ultrabeat pattern to the Arrange area: 1 Select the desired pattern in Ultrabeat’s Pattern menu. 2 Click-hold the area to the left of the pattern menu.
Playback Mode Pattern reactions to incoming MIDI notes is set in the Playback Mode menu. You will find the following options here: Â One Shot Trigger: The reception of a MIDI note starts the pattern, which plays once through its cycle, then stops. If the next note is received before the pattern has reached its final step, the new note stops playback of the first pattern and the next pattern begins playing immediately (this can be a different pattern or the same pattern, depending on the MIDI note received).
Creating Drum Sounds in Ultrabeat The following section covers a few specific sound creation tips. Please take the time to explore the vast and complex possibilities available to you in Ultrabeat, using the following programing tips as a starting point. You’ll discover that there is hardly a category of electronic drum sound that Ultrabeat can’t create easily. Note: In Ultrabeat’s Settings > Factory > Tutorial Settings folder, you will find a drum kit called Tutorial Kit.
3 Set the attack time in Env 1 to zero by sliding the leftmost of the two junction points that sit on the x-axis all the way to the left. 4 Experiment with the decay time by moving the rightmost of the two junction points that sit on the x-axis; you’ll discover that higher decay values (shifting the Bezier handle to the right) result in sounds similar to synth toms, while shorter decay values (shifting to the left) provide the kick character.
3 Vary the frequency of band 2 (easily recognizable in the blue part of the EQ graph) to influence the extent of bass drum tonality. A further method for reducing the tonality of a drum sound that is rich with overtones is to use a lowpass filter. In the following example, you will control the cutoff frequency of the filter with an envelope.
More Kick … To get even closer to the 909, use an EQ setting as shown in the following graphic. Note the low frequency pressure point around 60 Hz (which can be seen in the red area on the EQ graph) as well as the assertive punch or kick (the blue area starting at 460 Hz and up) of a 909 bass drum are strengthened. (This EQ setting is already part of the Kick 2 setting.) More Contour … In the example, all four envelopes are being used.
Note: The bass drum sound described is called Kick 3 in the Tutorial Kit at a pitch of D1. Use the second oscillator (with similar settings or with a sample) or use the filter and the ring modulator—the sky’s the limit as far as your imagination is concerned, so get on with it, and create that next “gotta have it” drum sound. Note: You can find an “emulation” of the legendary 808 bass drum under the name Kick 4 in the Tutorial Kit, at a pitch of D#1.
7 Use the filter parameters of the noise generator to either roughen up, refine, or add bright frequencies to the noise component of the snare drum sound. Select a LP filter type, and try a filter frequency between 0.60 and 0.90. Modulate it with LFO 1 that you’re already using to control the pitch of Oscillator 2. Note: The snare drum sound is called Snare 1 in the Tutorial Kit, at a pitch of E1. To refine the snare drum sound using FM synthesis: 1 Turn on Oscillator 1 in FM mode.
You have just very cleverly emulated both of the 808’s resonating filters. Shifting the pitch of both oscillators simulates the behavior of the 808’s Tone control by the way. To complete the 808 emulation by adding some noise: 1 Switch the noise generator on, and activate the highpass mode in its filter (HP). Set the Cutoff value to about 0.65, Resonance to 0.35 and add a little Dirt (around 0.06). 2 The noise generator provides the sustained snare sound.
You can now dynamically play the sound using velocity. To increase the performance dynamics: 1 Reduce the values of the individual volumes by turning down the Volume knobs in both oscillators and the noise generator. Note how the Mod ring and its Via sliders also move back.
4 Set the additional control that appears as shown below, to control the character of the sound with velocity: 5 Repeat this with the other parameters of Oscillator 2, as well as pitch: 6 Modulate the noise generator as follows: Â Cut parameter: Choose Max as modulation source, then set the modulation control as shown below. Â Dirt parameter: Choose LFO 2 as modulation source, then set the modulation control as shown below.
The Kraftwerk Snare A further classic electronic snare drum sound is the highly resonant lowpass filter of an analog synthesizer that quickly closes with a snap. This sound was used extensively by Kraftwerk. To recreate the Kraftwerk snare sound with Ultrabeat: 1 Select the Snare 1 sound. 2 Direct the signals of both oscillators and the noise generator to the main filter. 3 Modulate Cutoff with Env 1 (this is already modulating the volume of the noise generator). 4 Modulate the filter resonance with Env 2.
Creating Hi-Hats and Cymbals Electronic hi-hat sounds are very easy to create in Ultrabeat. To create a hi-hat in Ultrabeat: 1 Load the Standard Tutorial sound. 2 Switch off Oscillator 1 and turn on the noise generator. 3 Choose the following settings for the noise generator: In the screenshot above, you can see, that the Cutoff parameter is modulated by Env 1. The modulation is negative, the position of the Mod slider is below that of the base parameter value.
3 In the Material Pad of the Model oscillator, choose a setting with plenty of overtones as in the graphic below. 4 Set the volume of each oscillator to a value of –60 dB and turn the ring modulator on by clicking on its name. You’ve just created a bell-like sound that you can filter (with a high resonance value) if required. Note: You can find a similar sound called Ring Bell at a pitch of A2 in the Tutorial Kit.
30 GarageBand Instruments 30 GarageBand Instruments are automatically installed with Logic Pro. You can insert them as per other software instruments. GarageBand Instruments are software instrument plug-ins that are used in Apple’s GarageBand application. Their inclusion makes the importing of GarageBand files into Logic Pro a trouble-free experience. GarageBand Instruments are actually less CPU and memory-intensive versions of equivalent Logic Pro instrument plug-ins.
This has two main benefits: Â As the GarageBand instrument plug-ins are less CPU and memory-intensive, they load faster than the equivalent Logic Studio software instruments. Â Limitation to a few, powerful, parameters simplifies the use of the instruments. Play around with the parameters to see how easily spectacular sounds can be created! The macro parameter sliders available to each GarageBand Instrument are different.
Analog Mono This is a monophonic (one note can be played at a time) analog synthesizer lead sound. Unique macro parameters are: Â Glide: Determines the time it takes a note to change (slide) to another. Â Richness: Determines the complexity of the sound texture, making the sound fuller. Analog Pad The Analog Pad is based on the ES2. This is a warm analog synthesizer pad that is useful for a range of musical styles.
Digital Basic The Digital Basic instrument is based on the ES2. This is a basic digital synthesizer sound that is useful for a range of musical styles. Unique parameters are: Â Harmonics: Changes the sound dramatically as more harmonics (overtones) are added. The impact of this parameter is difficult to describe, so please experiment with it. Â Timbre: Changes the color of the sound from dark to bright. Digital Mono The Digital Mono instrument is based on the ES2.
Electric Clavinet The Electric Clavinet sound is based on the EVD6. It emulates the Hohner D6 clavinet. It offers the following unique parameter: Â Damper: Changes the tone of the clavinet, making it less sustained, and more woody sounding as you move towards the high setting. Electric Piano The Electric Piano sound is based on the EVP88. It sounds like the Fender Rhodes electric piano. Macro parameters are: Â Model: A more bell like tone is achieved when the Tines button is selected.
Piano, Sound Effects, and Strings The Piano, Sound Effects, and Strings sounds are sample-based. As with other GarageBand instruments, the Settings menu offers several variations. Tonewheel Organ The Tonewheel Organ sound is based on the EVB3. It emulates the Hammond B3 organ. Please try out the various Settings available, as this instrument is capable of generating a wide array of organ tones. Unique parameters are: Â Drawbars: Makes the sound a little thicker (more) or thinner (less).
Appendix Synthesizer Basics If you are new to synthesizers, you should read this chapter. It covers important facts about the synthesizer and explains the difference between analog, digital, and virtual analog synthesizers. Important synthesizer terms such as cutoff, resonance, envelope, and waveform are also introduced. Analog and Subtractive An analog synthesizer signal is an electrical signal, measured in volts.
Undesirable analog synthesizer phenomena, such as the habit of going completely out of tune, are not simulated by virtual analog synthesizers. You can, however, set the voices of the ES1 to randomly detune, adding life to the synthesizer’s sound.
Subtractive Synthesis Subtractive synthesis is synthesis using filters. All analog and virtual analog synthesizers use subtractive synthesis to generate sound. In analog synthesizers, the audio signal of each voice is generated by the oscillator. The oscillator generates an alternating current, using a selection of waveforms which contain differing amounts of (more or fewer) harmonics.
The picture below shows a sawtooth wave with the filter half closed (24 dB/Fat). The effect of the filter is somewhat like a graphic equalizer, with a fader set to a given cutoff frequency (the highest frequency being fed through) pulled all the way down (full rejection), so that the highs are damped. With this setting, the edges of the sawtooth wave are rounded, making it resemble a sine wave. The wave length here is not really higher, but the zoom setting is.
The effect of the resonating filter is comparable to a graphic equalizer with all faders higher than 660 Hz pulled all the way down, but with only 660 Hz (Cutoff Frequency) pushed to its maximum position (resonance). The faders for frequencies below 660 Hz remain in the middle (0 dB). If you switch off the oscillator signal, a maximum resonance setting results in the selfoscillation of the filter. It will then generate a sine wave.
Classic synthesizer literature encourages the use of the triangular wave for the creation of flute-like sounds. In the age of sampling, however, it’s pretty hard to sell a triangular wave as a flute sound to anyone. The screenshot above shows a rectangular wave. The rectangular wave contains all odd harmonics, the amplitudes of which decrease proportionately with their number. The pulse width can be set to any value and serves as a modulation address.
When you strike a key, the envelope travels from zero to its maximum level in the attack time, falls from this maximum level to the sustain level in the decay time, and maintains the sustain level as long as you hold the key. When the key is released, the envelope falls from its sustain level to zero over the release time. The brass or string-like envelope of the following sound—the envelope itself is not shown in this graphic—has longer attack and release times, and a higher sustain level.
Glossary Glossary AAC Abbreviation for Advanced Audio Codec. A compression and decompression algorithm and file format for audio data. AAF Abbreviation for Advanced Authoring Format. A cross-platform project exchange file format that you can use to import multiple audio tracks, inclusive of references to tracks, time positions, and volume automation. accelerando A gradual increase in tempo (see tempo).
ALAC Abbreviation for Apple Lossless Audio Codec, an encoding/decoding algorithm that delivers lossless audio compression. alias A pointer to a MIDI region in the Arrange area. An alias does not contain any data. It simply points to the data of the original MIDI region. You can create an alias by Shift-Option-dragging the original MIDI region to a new location. An alias can not be edited directly. Any change to the original region will be reflected in the alias.
audio file Any digital recording of sound, stored on your hard drive. You can store audio files in the AIFF, WAV, Sound Designer II (SDII), and CAF formats in Logic Pro. All recorded and bounced WAV files are in Broadcast Wave format. audio interface Device used to get sound into and out of your computer. An audio interface converts digital audio data, sent from your computer, into analog signals that speakers can broadcast.
binaural hearing A description of the way human beings process audio positioning information, allowing the direction of a signal source to be recognized (in front, behind, above, below, and to the left or right of the listening position). binaural panning A process that emulates binaural hearing. bit depth The number of bits used by a digital recording or digital device. The number of bits in each sample determines the (theoretical) maximum dynamic range of the audio data, regardless of sample rate.
checkbox A small box. You click a checkbox to select or deselect (or turn on/off ) an option. chorus effect Effect achieved by layering two identical sounds with a delay, and slightly modulating the delay time of one, or both, of the sounds. This makes the audio signal routed through the effect sound thicker and richer, giving the illusion of multiple voices. click Metronome, or metronome sound.
count-in Beats heard prior to the start of a recording (or playback). cueing Monitoring (hearing playback) while fast-forwarding or rewinding. cutoff frequency Frequency at which the audio signal passing through a low or highpass filter is attenuated by 3 dB. cutting The act of reducing a level, or frequency, when using EQ or other filters. Also used to describe physically dividing and removing sections of files, regions, and so on (see boosting and attenuation).
digital A description of data that is stored or transmitted as a sequence of ones and zeros. Most commonly, refers to binary data represented by electronic or electromagnetic signals. All files used in the Logic Studio applications are digital. Also see analog for comparison. Digital Full Scale See DFS. disclosure triangle A small triangle you click to show or hide details in the user interface.
EQ Shortened form of equalizer. Equalizers are used to boost or cut frequencies in an audio signal. There are several types of EQ available in the Logic Studio applications. equalization See EQ. event Individual MIDI command, such as a note on command. Continuous controller movements (modulation wheel, for example) produce a quick succession of individual events—each with an absolute value. expander An effect process that increases the dynamic range of an audio signal.
frequency The number of times a sound signal vibrates each second, measured in cycles per second, or Hertz (Hz). GM Abbreviation for General MIDI. A standard for MIDI sound modules that specifies a uniform set of instrument sounds on the 128 program numbers, a standardized key assignment for drum and percussion sounds on MIDI channel 10, 16-part multi-timbral performance and at least 24 voice polyphony. The GM specification is designed to ensure compatibility between MIDI devices.
instrument channel Logic Pro and MainStage support the use of software based instruments. Software instrument plug-ins are inserted into the Instrument slot of instrument channels. instrument object An object in the Logic Pro Environment designed to communicate with a single-channel MIDI device. An instrument object represents a physical or virtual device which handles MIDI information. Also see multi instrument object.
loop An audio file that contains recurring rhythmic musical elements, or elements suitable for repetition. LFO Abbreviation for Low Frequency Oscillator. An oscillator that delivers modulation signals below the audio frequency range—in the bandwidth that falls between 0.1 and 20 Hz, and sometimes as high as 50 Hz or 400 Hz. low cut filter A low cut filter is essentially a highpass filter that offers no slope or resonance controls. It attenuates all frequencies below the defined cutoff.
MIDI channel A MIDI channel is a “tube” for MIDI data, which flows through MIDI ports. Up to 16 separate MIDI channels can pass through a port simultaneously. Tracks recorded in Logic Pro can be directed to different tubes (channels), which can contain different information, and play back through different sounds, assigned to each channel. As examples, channel 1: piano, channel 2: bass, channel 3: strings and so on.
modulation Generally, a slight, continuously varying change. Many Logic Studio effects and synthesizers contain modulators. modulation amount The strength, or intensity, of modulation. modulation matrix The EXS24 mkII (and some other Logic Studio instruments) contains a grid that allows you to vary a number of target parameters, such as pitch, with a number of modulators (modulation sources). This grid is referred to as the modulation matrix in the EXS24 mkII.
normalize This function applies the current Parameter box settings to the selected MIDI events (by altering the actual events themselves), and clears existing parameter settings. When it comes to audio, a different Normalize function raises the volume of a recorded audio file to the maximum digital level, without altering the dynamic content. notch filter This filter type cuts the frequency band directly surrounding the cutoff frequency, allowing all other frequencies to pass.
plug-in Software application that enhances the functionality of the main program (in this case, one of your Logic Studio applications). plug-in window A window that opens when a plug-in is inserted, or the Insert/ Instrument slot is double-clicked. Allows you to interact with the plug-in parameters. post fader Sends in analog mixers are positioned either before (pre) or after (post) the fader.
region Regions can be found on Logic Pro Arrange tracks: They are rectangular beams that act as containers for audio or MIDI data. There are three different types of regions: audio regions, MIDI regions, and folder regions (usually referred to as folders). Also see: audio region, MIDI region, and folder. resonance A term generally associated with filters, particularly those of synthesizers. Resonance emphasizes the frequency range surrounding the cutoff frequency. See cutoff frequency.
saturation A term most commonly associated with a slight tape distortion or the characteristics of tube amplifiers. It basically describes a very high gain level that causes a slight distortion of the incoming signal, resulting in a warm, rounded sound. scale A group of related musical notes (or pitches) that forms the basis of the melody and harmony in a piece of music. The most common scales are the major scale and minor scale.
side chain A side chain is effectively an alternate input signal—usually routed into an effect—that is used to control an effect parameter. As an example, you could use a side chained track containing a drum loop to act as the control signal for a gate inserted on a sustained pad track, creating a rhythmic gating effect of the pad sound. single trigger mode This term is associated with synthesizers such as the ES1. In this mode, envelopes are not retriggered when tied (legato) notes are played.
step sequencer While all sequencers, including Logic Pro, step through a series of events, this term is used to describe a device from the seminal years of analog synthesizers. Essentially, two rows of knobs (usually 8) were individually adjusted to control the gate time (note length) and pitch of a connected synthesizer. The sequencer would step through these knob settings once, or repeatedly.
takes A take, put simply, is a recording. Logic Pro allows you to create several takes, one after the other, without leaving record mode. These takes can then be compiled into a super take (see comping). template A project that contains settings and preferences that you have defined. Templates serve as a starting point for new projects (scoring tasks, audio only projects, MIDI only projects, and so on, as your personal needs require).
VU meter Abbreviation for Volume Unit meter. An analog meter used to monitor audio levels. WAV, WAVE The primary audio file format used by Windows-compatible computers. In Logic Pro, all recorded and bounced WAV files are in Broadcast Wave format, which includes high-resolution timestamp information that stores positional information. This makes it easy to align these files in other audio and video applications. waveform A visual representation of an audio signal.
A Adaptive Limiter 61 additive synthesis 355 aftertouch event 629 AIFF file 629 AKAI file described 629 importing with EXS24 mkII 403–407 aliasing creating artificially 52 described 630 allpass filter 630 amplitude, described 630 amp modeling effects 19–25 Analog Basic (GarageBand Instrument) 616 Analog Mono (GarageBand Instrument) 617 Analog Pad (GarageBand Instrument) 617 analog signal 630 Analog Swirl (GarageBand Instrument) 617 Analog Sync (GarageBand Instrument) 617 analog synthesizer 621 attack phase
Clip Distortion 53 clipping 51, 633 colored noise 260 comb filter effect 141, 633 component modeling 19, 455 Compressor 62–65 Auto Gain parameter 65 Circuit Type menu 65 parameter overview 62 using 64 compressor described 59, 633 frequency-specific 65 controller, described 633 Controls view (plug-in) 633 convolution reverb 160, 169 Core Audio 633 Core MIDI 633 Correlation Meter 128 cutoff frequency 83, 624 D decibel 634 DeEsser 65 Detector section 66 Suppressor section 66 Delay Designer 28–45 adjusting fil
FM knob 232 FM parameters 232 Glide parameter 230 Harmonic knobs 231 LFO 233 Main Level knob 233 MIDI controller assignments 234 Modulation Env section 232 modulator parameters 231 Modulator Pitch knob 232 Modulator Wave knob 231 output parameters 233 Randomize parameter 230 Rate knob 233 Stereo Detune knob 233 Sub Osc Level knob 233 Transpose parameter 230 Tune parameter 230 Unison button 230 Velocity knob 234 Voices parameter 230 Vol Envelope knob 233 waveforms 231 Electric Clavinet (GarageBand Instrument
envelope trigger modes 288 Env Mode menu 297 Fat button 268 filter 262–270 cross-fading 263 mode 268 modulating 269 resetting 267 routing 262 slope 268 Filter Blend slider 263 Fix Timing button 299 Flanger button 300 FltBlend modulation target 277 Flt Reset button 267 FM (frequency modulation) 257 FM knob (Filter 2) 269 Frequency knob 255 Glide knob 252 global parameters 251–254 Hi button 268 Intensity knob 300 inv button (Router) 272 inverting via modulation 272 Legato button 252 Lfo1Asym modulation target
Env2Rel 278 Env2Time 278 Env3Atck 279 Env3Dec 279 Env3Rel 279 Env3Time 279 Glide 279 LFO1Rate 278 series filter routing 262 Sine Level knob 271 SineLevl modulation target 276 solo point (vector envelope) 296 Speed knob 300 Square 291 Stratocaster 315 Surround Diversity parameter 254 Surround Range parameter 254 Sustain and Sustain Time sliders 290 sustain point (vector envelope) 294 Sync option 259 target.
Korg CX-3 drawbar organ controller 349 Leakage slider 336 Leslie 344, 358 Lower Stretch slider 338 Lower Volume slider 329 LP Split parameter 326 Max Wheels slider 335 Mic Angle slider 346 MIDI CC parameter 334 MIDI controller assignment 348 MIDI Mode parameter 327 MIDI setup 324 MIDI to Presetkey parameter 333 Mode menu (reverb) 341 Mode menu (Sustain section) 340 Mode menu (Wah section) 342 Mode parameter (morphing) 334 morphing 334 Native Instruments B4D controller 352 NE MIDI mode 353 NI MIDI mode 352 O
Modulation section 375 Old D6 model 364 phaser 375 Picked model 365 Pickup Mode menu 372 Pickup Position section 371 pickup switches 362 Pitch Fall slider 370 Pluck model 365 Pressure parameter 362 Random slider 368 Range knob 374 Release slider 369 Shape slider 368 Sharp D6 model 364 Soft switch 362 Stereo Spread control 363 Stiffness slider 370 StrBells model 365 Stretch parameter 361 String section 369 Tension Mod slider 370 Tone knob 372 Treble switch 362 Tune parameter 361 Velocity slider 368 Velo Curv
Drive section 383 effects 383–386 emulated e-piano model Hohner Electra piano 389 Rhodes 387 Wurlitzer piano 389 EQ section 383 equal tempered scale 382 extended parameters 386 Gain knob 384 Intensity knob (Chorus section) 386 Intensity knob (Tremolo section) 385 Lower Stretch knob 382 MIDI controller list 390 Model dial 380 Model parameters 381–382 Phaser section 384 Rate knob (Phaser section) 384 Rate knob (Tremolo section) 385 Release knob 381 Stereo knob 381 Stereophase knob (Phaser section) 385 Stereop
Key Range parameters (group) 438 Key Range parameters (zone) 434 Key Scale parameter 419 Legato button 409 Level via Vel slider 418 LFO parameters 420 LFO waveform 420 Load Multiple Samples command 430 Loop parameters 435 lowpass filter slopes 416 Map Mod & Pitch Wheel to Ctrl 4 & 11 setting 441 modulation destinations 424 modulation matrix 421–425 modulation path 421 modulation sources 424 Mono button 409 multi output instrument 439 Options button 410 Pan parameter (zone) 434 Parameters window 408–425 Past
external audio effect, integrating in Logic Mixer 201 External Instrument 451 Input Volume 451 MIDI Destination 451 F Fat EQ 88–89 parameter overview 88 using 89 field effect transistor 56 filter 623 cutoff frequency 624 resonance 624 filter bank 207 filter effects 99–120, 636 filter slope 575, 636 flanger 300 Flanger (effect plug-in) 137 formant described 158 shifting 156 Fourier theorem 624 Freeze function 636 frequency band adjusting 83 analyzing 87 compressing 72 frequency modulation 229, 231–233, 257
Inspector, described 637 instruments included in Logic Pro 13 included in MainStage 18 interface, described 638 K Klopfgeist 453 L latency 638 legato 638 Leslie 358 level adjusting 200 analyzing 129 Level Meter (plug-in) 129 LFO 285, 639 Limiter (plug-in) 71 limiters 60 LinearPhase EQ 90 Link button, described 638 Logic Pro, included effects and instruments 11 loudness controlling 61 increasing 62 Low Cut Filter (effect plug-in) 96 lowpass filter 572 Low Pass Filter (effect plug-in) 96 Low Shelving EQ (ef
parameter overview 152 reference tuning 154 response time 154 Root field 153 Scale menu 153 pitch effects 151–158 Pitch Shifter II 155–156 parameter overview 155 using 156 PlatinumVerb 164–167 density and diffusion 166 early reflection parameters 165 High Cut parameter 166 Initial Delay parameter 166 low frequency band parameters 167 output parameters 166 Predelay parameter 166 reverb parameters 165 reverb time 166 plug-in window 643 post fader 643 predelay (reverb effect) 160 pre fader 643 Pulse Width Modu
Depth knob (morph envelope) 508 Depth via Vib Ctrl sliders 490 Env button (envelope display) 493 Env button (morph envelope) 506 envelope display 494 Envelope knob 488 envelope parameters 498–500 envelopes 493–500 adjusting curve shape 496 adjusting node position 495 comparing original and edited version 499 copying 496 determining sustain mode 498 editing 495 looping 498 modulation routing 494 nodes 495 recording 496, 546 scaling 498 syncing to project tempo 499 velocity sensitivity 498 zooming 495 extende
Morph X/Y menus 510 ms button (Envelopes section) 499 ms button (morph envelope) 508 Notch button 477 Note on Random section 491 object 457, 467–472, 513 activating 468 position 471 type 468 velocity sensitivity 468 object parameters 467–472 Off button (Level Limiter) 485 organ sounds 522 Output Width slider 479 Pad button (morph envelope) 506 Pad Mode menu (morph envelope) 506 parameter interactions 514 parameter overview 458 Peak button 477 percussive sounds 523 Phase knob (LFOs) 488 Phase knob (Vibrato s
Waveform menu (Vibrato section) 490 Waveshaper 475 Wet Level knob 478 Xfeed knob 478 self-oscillation 645 sequencer, described 645 sibilance, eliminating 65 Side Chain function, described 646 Silver Compressor 78 Silver EQ 97 Silver Gate 79 SilverVerb 167 sine sweep 202 single band EQs 83 single trigger mode 253, 460, 646 slope 83 Sound Effects (GarageBand Instrument) 620 SoundFont2 file, importing 399 Soundtrack Pro, included effects 14 Soundtrack Pro Autofilter 120 Soundtrack Pro Reverb 168 Space Designer
T Tape Delay 48–49 feedback 49 setting Groove value 49 shaping sound 49 tempo analyzing 128 described 648 Test Oscillator 202 threshold (compressor) 59 tick 648 time code, described 648 timing described 648 enhancing 193 timing problems, correcting 46 Tonewheel Organ (GarageBand Instrument) 620 tonewheel sound generation 358 tracking oscillator 107 transient analyzing 128 described 648 shaping 68 transposing audio signal 151, 155, 156 described 648 Tremolo (effect plug-in) 150 triangular wave 625 Tuned Perc
interface 555 kick drum 603 Kraftwerk snare 612 learning MIDI controller assignment 585 Length menu 592 Level knob 576 LFO 585 cycles 586 Ramp knob 586 rate 586 turning on/off 585 waveform 586 live performances 601 Material Pad 570 metallic sounds 613 MIDI Controller Assignment menu 585 Min/Max (Velocity) sliders 568 Model button 570 modulation 581–589 basic principle 581 highlighting target 589 Mod parameter 581 routing 584 Via parameter 581 modulation sources 585–589 multi output instrument 562 Mute butto
starting/stopping 592 Swing knob 592 switching to large grid display 597 triggering sound 594 turning on/off 591 Stiffness parameter 570 Swing button 593 Swing knob 592 Synthesizer section 563–580 toms 612 tonal percussion 612 Trigger menu 580 trigger row 594 Trigger Shortcut menu 594 tutorial 603–614 type buttons (Model section) 570 Vel Layer slider 568 velocity/gate row 596 Voice Auto Select function 557 Voice mode 591 Voice Mute Mode button 602 Voice Volume control 579 volume envelope 587 Volume fader 56
